an n avaya a solution n for r

AAnn AAvvaayyaa SSoolluuttiioonn ffoorr

Carlo De Luca Senior Proposal Manager

5606 Mira Montes Las Cruces NM 88007

908-953-5755 [email protected]

December 21, 2007

Presented by: Shirl Edwards

Senior Proposal Manager

409 Hannah Jeanne Circle New Smyrna Beach FL 32169


Alexander Lopez Senior Manager

Executive Briefing Program

8400 Doral Boulevard Miami FL 33166-6676


VoiceCon Request for Proposal for an IP Telephony System

Table of Contents

Table of Contents 1.0.0 System Overview......................................................................................................61.0.1 LAN/WAN Requirements.................................................................................................81.1.0 Commercial Availability and Customer References................................................91.1.1 Single System Image......................................................................................................101.1.2 Enhanced 911 (E911) Services Support ..................................................................121.2.0 Proposed IPTS .................................................................................................................141.3.0 IPTS Design Platform.....................................................................................................151.3.1 Common Control..............................................................................................................201.3.2 Call Processor Make/Model..........................................................................................251.3.3 Call Processing O/S .......................................................................................................261.3.4 Memory ............................................................................................................................... 261.3.5 Power Supply....................................................................................................................281.3.6 Ethernet Call Control Signaling Links........................................................................321.3.7 System Clocks..................................................................................................................341.4 Remote Survivability..............................................................................................341.4.1 Survivable IPTS Features/Services ...........................................................................361.4.2 Local Survivability Failover and Switchback............................................................381.4.3 Survivable Messaging Services ..................................................................................391.4.4 Network Failover Resiliency .........................................................................................391.5 Session Initiated Protocol (SIP) ..........................................................................401.6 Security.....................................................................................................................441.6.1 Unauthorized System Access......................................................................................441.6.2 Unauthorized Network Access ....................................................................................461.6.3 Disruption of Services ....................................................................................................471.6.4 Theft of Services ..............................................................................................................481.6.5 Restricted Calls ................................................................................................................491.6.6 Confidentiality and Privacy (Packet Sniffing) ..........................................................511.6.7 Physical Interfaces ..........................................................................................................521.6.8 Root Access ......................................................................................................................531.6.9 Miscellaneous Security Requirements......................................................................542.0 IPTS Network Port Capacity Requirements .....................................................562.1.0 Port Capacity Requirements ........................................................................................562.1.1 CHQ Facility ......................................................................................................................562.1.2 RO Facility .........................................................................................................................572.1.3 SO Facility..........................................................................................................................572.2 Port Requirements .................................................................................................572.2.1 CHQ Station/Trunk Port Requirements.....................................................................572.2.2 RO Station/Trunk Port Requirements........................................................................592.2.3 SO Station/Trunk Port Requirements........................................................................593.0 Port Interface and Traffic Handling Requirements.........................................613.1 Common Equipment..............................................................................................613.1.1 Universal Card Slots .......................................................................................................623.1.2 Common Equipment Redundancy .............................................................................623.2.1 IP Station Discovery........................................................................................................66

December 21, 2007 ©2007 Avaya Inc. Page i


Table of Contents

3.2.2 IP Station Power over Ethernet (PoE).......................................................................663.2.3 IP Station QoS..................................................................................................................673.4 VoIP Overflow Traffic.............................................................................................693.5.0 Port Interface Circuit Cards ..........................................................................................693.5.1 IP Telephones (desktop instrument and PC client softphones, including Attendant Console Position) & IP Audioconferencing Units.....................................................693.5.2 Analog telephones...........................................................................................................693.5.3 Facsimile terminal............................................................................................................713.5.4. Modem ................................................................................................................................ 723.5.5 Power Failure Transfer Station (PFTS).....................................................................723.5.6 GS/LS CO Trunk..............................................................................................................733.2.7 DS1/T-1 Carrier Interface Trunk..................................................................................743.2.8 Other Trunk Interfaces ...................................................................................................744.0.0. Voice Terminal Instruments.................................................................................754.1 Regulation Requirements.....................................................................................754.2 Desktop IP Telephone Instruments....................................................................754.2.1 Economy Desktop IP Telephone Instrument...........................................................764.2.2 Administrative Desktop IP Telephone Instrument .................................................774.2.3 Professional Desktop IP Telephone Instrument.....................................................794.2.4 Executive Desktop IP Telephone Instrument..........................................................824.2.5 ACD/Supervisor IP Telephone Instrument...............................................................844.2.6 Desktop IP Telephone Instrument Web Services Functionality ........................864.2.7 Desktop Instrument Options and Add-on Modules ...............................................874.2.8 SIP Compatibility..............................................................................................................904.2.9 Encryption ..........................................................................................................................904.3 Teleworker Options................................................................................................914.3.1 Desktop Teleworker ........................................................................................................914.3.2 Mobile Teleworker ...........................................................................................................934.4 Soft Attendant Console.........................................................................................944.5 IP Audio Conferencing Unit .................................................................................964.6 Mobile Extension ....................................................................................................974.6.1 Basic Mobile Cellular Extension Option....................................................................974.6.2 Advanced Mobile Client .................................................................................................994.7 Other IP Telephone Instruments.......................................................................1015.0 Call Processing Features ...................................................................................1025.1 Station User Features..........................................................................................1025.1.1 Additional Station User Features ..............................................................................1055.2 Attendant Operator Features.............................................................................1055.2.1 Additional Attendant Operator Features .................................................................1065.3 System Features...................................................................................................1065.3.1 Additional System Features........................................................................................109

December 21, 2007 ©2007 Avaya Inc. Page ii


Table of Contents

6.0.0 Systems Management .........................................................................................1106.0.1 System/Port Capacity...................................................................................................1126.0.2 Terminal Capacity..........................................................................................................1126.0.3 Support for Open Standards ......................................................................................1136.0.4 Security Features...........................................................................................................1146.0.5 User Interface & Tools .................................................................................................1146.1.0 Administration Functions .............................................................................................1246.1.2 Facilities Performance Management & Reports...................................................1256.2.1 Basic Trunk Usage and Traffic ..................................................................................1276.2.2 Attendant Consoles.......................................................................................................1296.2.3 Stations.............................................................................................................................1316.2.4 Traffic distribution ..........................................................................................................1326.2.5 Busy hour traffic analysis ............................................................................................1326.2.6 Erlang Ratings ................................................................................................................1326.2.7 Processor Occupancy ..................................................................................................1336.2.8 Threshold Alarms...........................................................................................................1336.2.9 Feature Usage................................................................................................................1346.2.10 VoIP Monitoring..............................................................................................................1356.3 Optional Reports ..................................................................................................1386.4.0 Call Detail Recording....................................................................................................1386.5.0 Maintenance....................................................................................................................1406.5.1 Alarm Conditions............................................................................................................1416.5.2 Maintenance Reports ...................................................................................................1416.5.3 Remote Maintenance ...................................................................................................1426.6.0 Provisioning .....................................................................................................................1437.0.0 Integrated Messaging System ...........................................................................1457.1.0 Support for Open Standards ......................................................................................1477.1.1 Security Features...........................................................................................................1487.2.0 Voice Mail Features..............................................................................................1507.2.1 Forwarding.......................................................................................................................1507.2.2. Disconnect Detection....................................................................................................1507.2.4 Answer Announcement................................................................................................1517.2.5 DTMF Signaling..............................................................................................................1527.2.6 Greeting ............................................................................................................................1537.2.7 Escape ..............................................................................................................................1537.2.8 Trunk Access ..................................................................................................................1537.2.9 Distribution Lists.............................................................................................................1547.2.10 Message Forwarding ....................................................................................................1557.2.11 Audit Trail .........................................................................................................................1557.2.12 Message Indication .......................................................................................................1567.2.13 Identification Code.........................................................................................................1567.2.14 Message Recovery .......................................................................................................1567.2.15 Message Reply...............................................................................................................1577.2.16 Message Review............................................................................................................1577.2.17 User Controls ..................................................................................................................157

December 21, 2007 ©2007 Avaya Inc. Page iii


Table of Contents

7.2.18 System Management Console ..................................................................................1587.2.19 Traffic Reports ................................................................................................................1607.2.20 System Changeability...................................................................................................1617.3.0 Networking.............................................................................................................1617.3.1 AMIS .................................................................................................................................. 1617.3.2 Digital IP Networking ....................................................................................................1617.4 Integrated Messaging Application....................................................................163Section 8: Unified Communications ........................................................................1658.0.1 UC System Integration.................................................................................................1668.0.2 Presence Management................................................................................................1728.0.3 Instant Messaging .........................................................................................................1738.0.4 UC Clients........................................................................................................................1748.0.5 Client Control ..................................................................................................................1788.0.6 Mobile Clients .................................................................................................................1868.0.7 Microsoft Integration .....................................................................................................1948.0.8 IBM Integration ...............................................................................................................1978.1 Conferencing & Collaboration...........................................................................1978.1.1 Ad Hoc Conference.......................................................................................................1978.1.2 Scheduled Conference ................................................................................................1978.1.3 Host Control Features ..................................................................................................1988.1.4 Web Collaboration.........................................................................................................1998.1.5 Application Sharing .......................................................................................................2008.2 UC Architecture ....................................................................................................2008.3 Systems Management .........................................................................................2048.4 Security...................................................................................................................208Section 9: Automatic Call Distribution (ACD) Contact Call Center ..................2119.0 Basic ACD Call Center Requirements .............................................................2119.0.1 Third Party System Integration ..................................................................................2129.0.2 ACD Station Equipment...............................................................................................2139.0.3 Remote ACD Station Equipment ..............................................................................2199.0.4 Redundancy ....................................................................................................................2229.1 ACD Contact Call Center Parameters..............................................................2249.2 Basic ACD Features.............................................................................................2259.3 Call Flow .................................................................................................................2279.3.1 Routing & Queuing........................................................................................................2279.4 Call Processing Functions .................................................................................2299.5 Supervisor Functions..........................................................................................2369.6 Agent Functions ...................................................................................................2429.7 System Call Prompts & Announcements .......................................................2549.7.1 Hands Free Caller Prompt Response .....................................................................2559.8 MIS Reporting System.........................................................................................2589.8.1 System Requirements..................................................................................................2599.8.2 MIS Reports.....................................................................................................................2619.8.3 Real Time Reports ........................................................................................................2629.8.4 Historical Reports ..........................................................................................................265

December 21, 2007 ©2007 Avaya Inc. Page iv


Table of Contents

9.8.5 Customized Reporting..................................................................................................2669.8.6 Terminal Report Access ..............................................................................................2749.8.7 Scheduled and Email of Reports ..............................................................................2749.8.8 Report Formats ..............................................................................................................2749.9. ACD Management & Administration ................................................................274

Part 2: System Pricing ........................................................................................ 2761.0 System Pricing Requirements...........................................................................2762.0 Summary Pricing – VoiceCon IPTS solution .................................................2762.2 Software Upgrade Subscriptions......................................................................2763.0 Desktop Voice Terminal Pricing .......................................................................2774.0 Detailed Configuration Components and Pricing.........................................278

Avaya Appendices

Avaya Appendix-1 Communication Manager 5.0 Features

Avaya Appendix-5 Avaya Contact Center Offers

Avaya Appendix-6 Avaya CMS Reports

Avaya Appendix-8 Avaya Call Center Vectoring

Avaya Appendix-9 Avaya Contact Center Patents

Avaya Appendix-10 Avaya Call Center Software Call Handling Options Review

Avaya Appendix-2 Standard-Optional Features Communication Manager

Avaya Appendix-3 PPT-Slides

Avaya Appendix-4 Voice Con08 Propriety References

Avaya Appendix-7 UC Client Info

December 21, 2007 ©2007 Avaya Inc. Page v

VoiceCon 2008 Request for Proposal for an IP Telephony System

Part 1 - Section 1 - System Overview

PART 1: System Performance Requirements Submit Part 1 responses in MS Office WORD file format except when otherwise noted. Avaya Response:

Comply.

1.0.0 System Overview VoiceCon Company (VoiceCon) plans to install a new IP Telephony System (IPTS) at its newly constructed Campus Headquarters (CHQ) facility. The IPTS will also support two remote facilities, a Regional Office (RO) and a Satellite Office (SO).

Distributed IPTS common control equipment must be installed in two equipment rooms located across the CHQ facility. All proposed call telephony servers must independently support all generic software features for the proposed IPTS as required in Section 5 of this RFP. The RO and SO facilities will have local trunk services, but will be configured as survivable remotes behind the centrally located IPTS at CHQ. It is mandatory that a single system image IPTS solution be proposed that satisfies the requirements listed later in this section of the RFP.

The proposed IPTS must support 1,500 stations users distributed across the CHQ, RO, and SO facilities at time of system cutover. Anticipated VoiceCon expansion plans will require the proposed IPTS be capable of supporting 50% growth of all call processing and port capacity parameters distributed proportionately across the three facilities. The proposed IPTS must support this growth requirement without replacement of any installed hardware or generic software (excluding new release versions).

The IPTS network configuration can be seen in Figure 1:

CHQ

RO 200 Stations

SO 50 Stations

WAN

Voicecon IPTS Network

CHQ: Campus Headquarters RO: Regional Office SO: Satellite Branch

Equipment Room 1

750 Stations

Equipment Room 2

500 Stations

Figure 1

December 21, 2007 ©2007 Avaya Inc. Page 6



VoiceCon plans to install a LAN/WAN cabling and a transport infrastructure that will fully satisfy the stringent requirements of IP Telephony communications for all intra-premises and inter-premises call control and voice communications transmissions. Each location will be equipped, at minimum, with a 1-Gbps Ethernet backbone. The local wiring closets will house 10/100/1000 Mbps Ethernet switches equipped with Power over Ethernet (PoE). Multi-service routers will be installed at all locations to support a MPLS WAN installation. All Ethernet switches and IP WAN routers will be equipped and programmed to satisfy QoS and security standards to support voice communications quality acceptable to VoiceCon. Pertinent bandwidth, latency, packet loss, and echo issues will be addressed by the LAN/WAN design and implementation.

Each station user’s work area will be supported by four (4) four-pair, Category 5E cable wiring with one (1) RJ-11 wall connector and three (3) RJ-45 wall connectors to the local wiring closet. The RJ-11 and RJ-45 connectors will be either wall mounted or mounted in the modular furniture throughout the office environment. NOTE: The proposed IPTS will be required to support a limited number of non-IP stations, e.g., analog telephones, requiring a RJ-11 connector. The proposed system may use circuit switched port carriers, LAN-connected media gateways, or some combination to support IPTS desktop analog communications and PSTN connectivity requirements.

Vendor Response Requirement

Based on the RFP requirements prepare a simple network diagram that illustrates the proposed IPTS design. Include in the diagram the brand name/model of the IPTSs, circuit switched port carrier/media gateway equipment, the brand/name of the IPTS systems management and messaging system. The diagram must be prepared and submitted in a separate attached file using MS PowerPoint format, and also copy/paste the diagram in the MS WORD submission.

Avaya Response:

Comply; please also see Avaya Appendix 3- PPT Slides 1 through 4 for additional drawings.




1.0.1 LAN/WAN Requirements VoiceCon has not yet decided on the make/manufacturer of its new LAN/WAN communications equipment.


Indicate if the proposed IPTS solution is requires manufacturer-specific LAN/WAN communications equipment to support any or all of the following voice communications operations or functions: call processing, port interface, network switching and routing, PoE, media gateway, QoS and security. Identify make/models of manufacturer-specific equipment if required.

Avaya Response:

The proposed Avaya Communication environment is open and standards based – essentially network infrastructure agnostic, for real-world flexibility. In the complex world of IP Telephony, we understand that enterprises often “inherit” LAN/WAN infrastructures from multiple vendors via mergers and acquisitions. Avaya has talented engineers in-house who are qualified to assess, evaluate, and recommend solutions that will protect your current infrastructure investment as well as support future requirements. We recommend that every client planning to install an IP Telephony solution have a Network Assessment and Recommendation engagement with our Avaya Global Services Organization.




Avaya has done successful interoperability testing with several major LAN/WAN infrastructure vendors, including Cisco, Nortel Networks, Juniper, and Extreme Networks. More information on this and additional system testing is available upon request.

1.1.0 Commercial Availability and Customer References The proposed IPTS equipment should be in current production and operating as part of a commercial system at a minimum of five (5) different customer installations.


State if the proposed IPTS equipment satisfies this commercial availability requirement. If the IPTS model has not yet been shipped and installed in a commercial installation, state expected availability date.

NOTE: All proposed system hardware and software must be formally announced as of VoiceCon Orlando 2008 to be accepted by VoiceCon as a response to this RFP. This is a mandatory requirement.

Avaya Response:

Comply with clarification. The proposed solution is in production and operating for numerous customers throughout the USA and Internationally. We have provided corporation names and the other requested information here, however, individual contact names and telephone numbers have been provided to Allan Sulkin in a separate PROPRIETARY – RESTRICTED

document, due to the web posting and mass distribution of this proposal. Names and telephone numbers of individual contacts are available on an as needed basis. Please contact your Avaya Client Executive or send an email to [email protected] for additional information. For additional information please see Avaya Appendix 4, References.

Reference Companies:

Customer Name Date of Installation

Current System Release

Western Kentucky University

2007 Avaya Communication Manager, Release 3.1 over 7 campuses.

Career Education Corporation (CEC)

Kinetic Concepts

Kansas State University

2003

2006

2006

Avaya Communication Manager, Release 3.0 over 5 locations Avaya Communication Manager, Release 3.0 over 5 locations Avaya Communication Manager, Release 3.x over 20,000 endpoints.

Virginia Commonwealth University (VCU)

2007 Avaya Communication Manager, Release 3.1 over 120 campus buildings




1.1.1 Single System Image The proposed IPTS must provide to all system subscribers and administrators a Single System Image across VoiceCon CHQ, RO and SO facilities. The Single System Image must include, but not be limited to, the following:

1) 4- or 5-digit dialing between all station users;

2) Virtually 100% transparent operation across all VoiceCon facilities and buildings of station, attendant, and system features (see RFP Section 5: Call Processing Features);

3) CHQ-housed systems management server(s) utilizing a single unified database for all station user profiles, equipped system design, and system-level operations;

4) Network-wide attendant operator services across all VoiceCon facilities and buildings, including the capability to support a centrally located attendant pool;

5) CHQ-housed shared messaging system resources;

6) Automatic alternative routing across the network for all voice calls (station-to-station and PSTN trunk connections).

Vendor Response Requirement: Answer each of the following questions:

1. Is the proposed IPTS network solution a true single system solution or multiple systems intelligently networked?

2. Does the proposed IPTS network solution fully satisfy all six (6) of the stated Single System Image requirements? If not, identify and explain which of the requirements are not fully satisfied?

Avaya Response:

Comply. The proposed solution is a single system, controlled by a centralized S8500C Server with additional S8500C Servers running as Local Survivable Processors (LSP) in CHQ1 and in CHQ2.

The solution supports the six Single System Image requirements as follows:

1. 5-digit dialing – since one centralized server controls the design, all extensions will be assigned and controlled by the S8500C Server at the CHQ Equipment Room 1 location. There will also be an additional S8500C server running in LSP mode in the unlikely event of a server failure.

The Campus Headquarters Equipment Room 2 will contain an additional S8500C Server running in Local Survivable Processor (LSP) mode; each individual server will be able to run the Enterprise individually or act as independent servers to control each physical location, if the WAN were to fail (up to 2400 stations and 800 trunks per S8500 purchased at the HQ location).

The Regional Office (RO) as well as the Satellite Office (SO) will have an S8300 Local Survivable Processor (LSP) that can run independent during a network failure or a CHQ catastrophic failure, they will obtain translations (programming) from the CHQ S8500 Server. The S8300 LSP locations can support up to 450 IP endpoints if required during a prolonged outage




2. A major advantage of the proposed solution is the ability of the locations to have the same extension at multiple sites. For example, if associates travel between locations, it would be convenient for the Mail Room and/or Guard’s desk to have the same extension number at all locations. The Multi-location dialing feature of Avaya Communication Manager supports this functionality. In addition, if the branch locations are currently autonomous, there may be duplicated extension numbers at several locations. Multi-location dialing minimizes user disruption by minimizing the extension numbers that have to be changed.

3. A High Level of Transparent Operation for Commonly Used Station, Attendant, and System Features – since the solution is a single system, transparent operation and features is an inherent capability.

Centralized Systems Management and Maintenance Operations Using a Single Unified Customer Database for All Location Equipment and Station Users – the proposed solution includes Avaya Network Management. This allow users to access and manage the system from anywhere on the network, or a remote location. Users will have access to any elements of the entire network, assuming they have the appropriate access credentials.

4. Avaya recommended attendant services provides the required capabilities for a centralized attendant operation (pooled or non-pooled) which supports all locations without loss of feature functionality; including station busy indication, call diversion, call status, emergency access, call display, call transfer and more.

5. Shared Messaging System Resources – the proposed solution includes Avaya Modular Messaging for all stations on the network. All users will have exactly the same features, capabilities, and access.

6. Automatic Alternative Routing (AAR) across the Network for IP and Circuit Switched Trunk Calls – Avaya Communication Manager supports many routing options including AAR, which will all be considered and implemented according to VoiceCon’s requirements at installation.




1.1.2 Enhanced 911 (E911) Services Support It is mandatory that the proposed IPTS support E911 services provided by a public safety answering point (PSAP) as defined by FCC regulations. Each of the three VoiceCon facilities utilizes a different PSAP.

All VoiceCon IPTS station user E911 calls must be directed to their local PSAP for call handling and response regardless of location, VoiceCon corporate facilities and teleworker residence. If more than one E911 solution is available for the proposed IPTS network configuration clearly specify the solution that is included in the price proposal.

Vendor Response Requirement:

Confirm that the proposed communications system solution supports E911 service for all user stations (IP and analog) regardless of physical location. Also, briefly explain how E911 service requirements are supported, specifically addressing each of the following questions:

1. A description of any optional hardware/software equipment, including peripheral servers. [Note: Include the price of all required equipment, including servers, in the pricing proposal]

Avaya Response:

E-911 capabilities are included with the latest release of Avaya Communication Manager at no additional cost - no additional servers are required.

However with IP telephones, users can move their phones from one location to another without telling their switch administrator. The E911: ELIN by range of IP addresses feature of Avaya Communication Manager allows the public service personnel to locate the 911 caller even if the caller neglected to inform the switch administrator that he/she moved the IP phone. This feature works for both types of phones: H.323 and SIP.

The E911 ELIN for IP wired extensions feature automates the process of assigning an emergency location information number (ELIN) through an IP sub network (“subnet”) during a 911 call. The ELIN is then sent over either CAMA or ISDN PRI trunks to the emergency services network when a 911 call is dialed. This feature properly identifies locations of wired IP telephones that call an emergency number from anywhere on a campus or location.

RedSky E911 Manager runs on a server running Windows 2003 Server.

E-911 Manager System Requirements

The server to support E-911 Manager, will be supplied by VoiceCon. The System Requirements are:

Database/Task Server Client Workstations

Hardware/Software Recommended

Minimum Hardware/Software Recommended Minimum

Processor Pentium IV, 1 Ghz Processor(s) Pentium III, 500 Mhz

RAM Memory 1 GB RAM Memory 128 MB

Hard Disk (free space) 8 GB, SCSI, RAID 5 Hard Disk (free space) 2 GB

Monitor 15” SVGA Monitor 17” SVGA, 800 x 600

Network Adapter 100 Mbit Network Adapter 100 Mbit




Database/Task Server Client Workstations

Hardware/Software Recommended

Minimum Hardware/Software Recommended Minimum

Keyboard 101 Enhanced Keyboard 101 Enhanced

Mouse Standard 2-button Mouse Standard 2-button

CD-ROM Yes (for s/w install) CD-ROM Yes (for s/w install)

2. How are station user moves/adds/changes reported to the PSAP?

Avaya Response:

This is a completely automated process by E911 Manager. If moves/add/changes are done in the IP PBX administration software, these changes are automatically picked up by E911 Manager and new ALI records are forwarded to the local exchange carrier (LEC) ALI DB that feeds the PSAP. If a new ERL/ELIN or network region/ELIN is created in the IP PBX administration software or in E911 Manager itself, E911 Manager creates and forwards new ALI records to the LEC ALI DB that feeds the PSAP Comply

3) What degree of specificity station user location is identified to the E911 PSAP? Desktop work area, local switch room, work floor, other?

Avaya Response:

E911 Manager supports all of the above. Location can be as granular as the desktop using Layer 2 port level network discovery or station screen administration. The Network Region/IP range approach is typically used to identify location to the floor or quadrant (NE, NW, etc) of a floor.

All 911 calls will be sent to the appropriate local PSAP. All IP, Digital and IP phones are supported.

1.1.2.1 E911 and Station Moves Station user moves behind the proposed IPTS solution should be tracked dynamically in real time for E911 services support.


Indicate if the proposed E911 solution satisfies this requirement and indicate how often the local and PSAP databases are updated.

Avaya Response:

Comply – The proposed E911 solution satisfies the requirement, all station user moves behind the IPTS solution are tracked dynamically in real time.. E911 Manager can be programmed to update local exchange carrier (LEC) ALI databases as a scheduled task. (Time range: once very 5 minutes to once every 24 hours) However, since the LEC ALI database only updates itself once every 24 hours, most customers update the local LEC ALI database once a day.




1.2.0 Proposed IPTS Vendor Response Requirement

Provide the following information regarding your proposed IPTS as details are requested in following sections:

1. Product and model name(s) for the IPTS and messaging system

2. Software release for each product/model proposed

3. Product/model commercial availability dates

Avaya Response:

Comply.

Please refer to the diagram provided in Avaya Appendix 3, PowerPoint Illustrations slides 2 through 6.

The Proposed IP telephony system is the Avaya S8500C Server, powered by Avaya Communication Manager, Release 5.0 and includes an entitlement for Avaya Integrated Management, Release 5.0

The messaging system is Avaya Modular Messaging with Message Storage Server (MSS) Release 3.1 with one-X Speech Release 2.1

Avaya Call Center Elite features are inherent to Avaya Communication Manager 5.0. Realtime and historical MIS reporting are supported using Call Center Elite and Avaya Call Management System (CMS) R14.

The proposed solution is a single system design with the centralized common control S8500C Server installed at CHQ1 location: all call processing operations for Branch/Remote locations are dependent on the CHQ1 location except in survivable mode. The call processing is centralized.

The Media Gateways used to support the station/trunk call processing and switching operations are listed below:

Location Media Gateway

CHQ Equipment Rm. 1 Two Avaya G450 Media Gateways running off a co-located S8500C Media Server with (1) S8500C Server as a Local Survivable Processor (LSP)

CHQ Equipment Rm. 2 Avaya G450 Media Gateway running off the HQ1 S8500C Server with (1) S8500 Server as a Local Survivable Processor (LSP)

RO Avaya G450 Media Gateway running off the HQ1 S8500C Server with S8300 Server as a Local Survivable Processor (LSP)

SO Avaya G350 Media Gateway running off the HQ1 S8500C Server with S8300 Server as a Local Survivable Processor (LSP)




Dates the products were first released are listed below:

Product Date of First Date of Proposed Release Release

Avaya Communication Manager May 6, 2002 January 7, 2008

Avaya Modular Messaging January, 2003 February 5, 2007

Avaya S8500 Media Server December 8, 2003 February 6, 2006

Avaya S8300 Media Server May 6, 2002 February 28, 2007

Avaya G450 Media Gateway January 2008 January 7, 2008

Avaya G350 Media Gateway December 8, 2003 March 5, 2007

Avaya UC (Unified Communications) Standard Edition

May 21, 2007 May 21, 2007

Avaya Call Management System December 8, 2003 March 19, 2007

Avaya Integrated Management May 6, 2002 January 7, 2008

1.3.0 IPTS Design Platform The proposed system solution may be based on either of the following two architecture system designs:

• Converged TDM/IP: call telephony servers supporting LAN/WAN distributed circuit switched port interface cabinets with integrated media gateway interfaces for IP port connectivity

• Client/server: call telephony servers supporting LAN-connected media gateway equipment for non-IP port connectivity

Vendor Response Requirement: Briefly describe the system architecture and design elements of the proposed IPTS solution. Include in your system description basic information about the following common equipment hardware elements:

1. Architecture design: converged or client/server)

2. Call telephony servers and associated common control equipment

3. Circuit switched port carrier/interface equipment, if applicable

4. LAN-connected media gateways (server-embedded, standalone, switch/router-equipped, desktop), if applicable

Avaya Response:

Comply. The proposed solution is a combination of the Converged and the Pure Client Server.

The CHQ and RO sites will have the G450 Media Gateways that will house the same signaling interface cards (media modules) as the SO and the G350 Gateway. Media Modules can be configured as needed to support analog or digital stations and trunks communication to the server is via IP.




A brief description of each Media Gateway type and the associated circuit packs or media modules is provided below:

New Avaya G450 Media Gateway

The G450 Media Gateway is a media gateway that can be deployed in medium to large sized branch locations or in wiring closets that service buildings and floors in a campus environment. It works in conjunction with Avaya Communication Manager 5.0 software running on Avaya S8X00 Servers to help deliver intelligent communications to enterprises of all sizes.

The Avaya G450 Media Gateway Chassis

The G450 combines telephony and data networking by providing PSTN toll bypass and routing data and VoIP traffic over the WAN. The G450 provides full support for Avaya IP and digital telephones, as well as analog devices like modems, fax machines and telephones.

A single G450 can support up to 450 users when deployed with an S8300 as a branch gateway in a mid to large branch office or Enterprise. It can serve up to 2400 users when deployed with S8500 as a campus gateway or in a large branch office.

The G450 can be configured as a standalone S8300 Server or as a survivable gateway to an Avaya S8500 Server operating as an External Call Controller (ECC). As a survivable gateway, the G450 can act as a Local Survivable Processor (LSP) to provide full featured telephone service to the gateway or it can provide basic telephony features in the Standard Local Survivability (SLS) mode.

The G450 is a scalable device with a basic configuration consisting of 1 Power Supply Unit (PSU), 1 Main board and a single Digital Signal Processing (DSP) child board supporting 80 or 20 VoIP channels. This configuration can be enhanced by adding a redundant PSU unit and up to three additional DSP child boards, increasing the number of VoIP channels up to 240 channels.

The G450 is a modular device, adaptable to support different combinations of endpoint devices. While fixed front panel ports support the connection of external LAN switches, network data ports, Ethernet WAN lines and external routers, eight slots are provided for plugging in optional media modules.

Pluggable media modules provide interfaces for different types of telephones, trunks, and WAN links. A combination is selected to suit the needs of the branch. Other telephony modules are also available to provide full support for legacy endpoints such as analog and digital telephones. A range of WAN modules provide support for USP and E1/T1 WAN links. IP phones are supported via an external LAN switch.

The G450 chassis features field replaceable DSPs, PSUs, fan tray, and main board for enhanced reliability.




Supervisor - Field removable motherboard for the G450

¾ 2 Triple Speed Ethernet LAN interface

¾ 2 Fast Ethernet WAN interface

¾ Contact closure Adjunct

¾ ETR port

¾ Console and Services ports

¾ 2 USB ports (for modem and DOK)

¾ Compact Flash slot

For more information on Branch Solutions or the G450 Media Gateways, please visit www.avaya.com and locate G450 Media Gateway under Products A-Z.

Avaya G350 Media Gateway

The G350 Media Gateway is a complete branch office communications solution ideally suited for branch offices with 8-72 extensions. This gateway allows organizations to economically extend their headquarters communications applications such as the Avaya Communication Manager out to their branch locations to lower overall costs and increase collaboration across the entire organization.

This is a powerful converged networking solution that packs an IP telephony gateway, an advanced IP WAN router, a VPN Gateway and a high-performance LAN switch into a compact (3U) modular chassis. Designed to be a complete voice/data networking solution, the G350 Gateway is ideally suited for enterprises with distributed branch office locations using 8-72 extensions. Enterprises benefit long-term by leveraging existing telephony investments while supporting an evolution to recent protocols such as H.323 or SIP-based telephony.

An advanced TDM/IP architecture provides seamless connectivity and communications between a wide variety of analog, digital, H.323, and SIP-based telephony devices and applications. For communications security, the G350 can secure VoIP media streams using Advanced Encryption Standard (AES), approved for use by U.S. Government agencies to protect sensitive information.

The Avaya G350 is also an edge router designed to support the consolidation of voice and data traffic over IP. Available IP WAN routing media modules add support for PPP/Frame Relay connectivity over E1/T1 or Universal Serial Port (USP) interfaces. Alternatively the G350 media gateway can connect to an external WAN device via a fixed 10/100 Ethernet WAN router port, which supports traffic shaping to match data transfer rates with available WAN bandwidth.




Media Modules

Analog Stations and Analog Trunks

Both the G450 and G350 Media Gateways support analog stations or trunks via the MM711, 8 port Media Module.

MM711 Analog Media Module

The MM711 provides the administrator with the capability to configure any of the eight ports of this analog circuit pack as:

¾ A loop start or a ground start central office trunk

¾ Loop current 18-60mA

¾ A wink start or a immediate start Analog Direct Inward Dialing (DID) trunk

¾ A 2-wire analog Outgoing CAMA E911 trunk, for connectivity to the PSTN

¾ MF signaling is supported for CAMA ports

¾ Analog, tip/ring devices such as single-line telephones with or without LED message waiting indication

The MM711 also supports:

¾ Type 1 and Type 2 Caller ID

¾ Ring voltage generation for a variety of international frequencies and cadences

¾ A hard-wired ground wire is added for each IROB-to-earth ground

Avaya MM716

24-Port Analog Media Module

The MM716 Analog Media Module is a 24 port analog circuit pack. All ports are dedicated for station (line). The station (line) ports support an LED Message Waiting Indicator, and are used for DID trunk (wink or immediate start) port connections. The MM716 is supported in the Media Gateways. Communication Manager 3.0 or higher, and equivalent gateway software, is necessary to support the MM716.




Digital Trunks

The Media Gateways use a MM710 E1/T1 Media Module to digital trunks. The MM710 has a built-in Channel Service Unit (CSU) so that an external CSU is not necessary. See the following illustration for an example of the MM710.

Avaya MM710 T1/E1 Media Module

Highlights of the MM710:

¾ Software selectable T1 or E1 operation

¾ An integrated CSU

¾ Both A-law (E1) and µ-law (T1) gain control and echo cancellation ability

¾ D4, ESF, or CEPT framing

¾ ISDN PRI capability (23B + D or 30B + D)

¾ AMI, ZCS, B8ZS (T1) or HDB3 (E1) line coding

¾ Trunk signaling to support US and international CO or tie trunks

¾ Echo cancellation in either direction

¾ Fractional T1 support

¾ An OIC DB 25-pin interface

¾ A Bantam loop back jack that is used for testing of T1 or E1 circuits.

The MM710 supports the universal DS1 that conforms to the ANSI T1.403 1.544 Mbps T1 standard and to the ITU-T G.703 2.048 Mbps E1 standard.

The Avaya Portfolio at a glance is run with Communication Manger Software 5.0 is the latest version GA in January 2008. The Media servers are chosen based on capacities and reliabilities. Media Gateway and IP devises are chosen based on VoiceCon’s preferences.

For customers who intend to grow over the 2250 IP end points VoiceCon has requested Avaya has a wide variety which would meet the needs of any small, medium largeenterprise. In fact the S8730 proposed would be able to support 12,000 IP endpoints.

All adjuncts would be consistent across the entire product line so we have built ininvestment protection, in fact if VoiceCon would outgrow this particular design they could then can add additional S8730 Media Servers to this or any another site, and network them to support close to one million endpoints.




Part of the Avaya IP Telephony portfolio

1.3.1 Common Control There are several mandatory common control requirements.

1.3.1.1 Common Control Housing VoiceCon requires that all IPTS host common control elements be fully embedded in compact housing with internal interfaces to media gateways, non-IP port circuit interfaces, and service circuit boards. Call control signaling to/from all IP endpoints must be supported through an integrated Ethernet LAN uplink connector, e.g., RJ-45. The cost benefits of this fully integrated design are reduced hardware, power, and system footprint requirements.

Avaya Response:

Read and understood




1.3.1.1.1 CHQ Host Common Control

Vendor Response Requirement: Briefly describe the hardware housing for all CHQ-located common control elements (call processing, signaling, et al), specifically including size (H x L x W), weight (standard common assembly), fan cooling units, and all embedded hardware components. Include in the response technical information for both CHQ-located host common control and RO/So-located local survivable processing equipment.

Avaya Response:

At CHQ1 Avaya proposes an S8500C Server running Communication Manager 5.0 with an additional S8500C Server running in LSP mode. At the CHQ2 equipment room there is an additional S8500C Server running Communication Manager in LSP mode this allows for a distributed enterprise over an IP infrastructure ideal for the mid-size or large business (up to 3200 ports each).

Avaya S8500C Server

The S8500C Server brings all of the robust performance and rich feature set of Communication Manager onto an industry standard processor running an industry standard operating system. This processing engine drives the switching fabric of the system. The S8500C Server is a Pentium 4 server running Linux operating system. It has 512MB of memory and a high-reliability (non-redundant) auto-sensing power supply. The server has a compact flash and reader. The simplex S8500C Server is 1 U high and fits into a 19" fourpost data rack.

The Processor Ethernet (PE) interface provides the connectivity between the gateways/IP endpoints/IP adjuncts and the Communication Manager without the traditional CLAN PN.

Parameter S8500C Specifications

Acoustical Noise Emissions

Sound power, idling: 65 decibel maximum Sound power, operating: 65 decibel maximum

Environment: Media server on: Air 50.0°F to 95.0°F (10°C to 35°C) Temperature Altitude: 0 ft to 2999 ft (0 m to 914 m) Media server off: -40°F to 140°F

(-40° to 60° C) Maximum altitude: 6998 ft (2133 m)

Environment: Humidity

Media Server on: 8% to 80% Media Server off: 8% to 80%

Heat Output Minimum configuration: 341 BTU (100 W) Maximum configuration: 1024 BTU (300 W)

Electrical Receptacle U.S.: NEMA 5-15 A




Parameter S8500C Specifications

Input Circuit Breaker: 15 A Sine-wave input (47 Hz to 63 Hz) required Input voltage low range: 100 – 127 VAC Input voltage high range: 200 – 240 VAC Input kilovolt-amperes (approximate):0.10 – 0.55 kVA Amp draw: 100 to 127 V ~ 6.0 A 200 to 240 V ~ 3.0A

If VoiceCon determines that SIP Phones are required an optional SIP Enablement Server (SES) is required which has the same environmentals as the S8500C.

1.3.1.1.2 RO/SO Local Common Control

Vendor Response Requirement: Briefly describe the hardware housing for all RO and SO local common control elements (call processing, signaling, et al), specifically including size (H x L x W), weight (standard common assembly), fan cooling units, and all embedded hardware components.

Avaya Response:

The RO and SO will be running the S8300 Server in an LSP mode. The S8300 Server with a G450 or G350 Media Gateway provides a flexible solution for growing companies seeking to streamline voice and data operations over one networked infrastructure. It can be used as a standalone solution for a small business or as a remote edge device for a larger enterprise network ranging between 8 to 450 stations.

The S8300 Server resides inside a G700, G450, G350 or G250 Media Gateway can be used as a standalone communication system or as a fully survivable remote gateway. The S8300 can be the primary controller for up to 50 remote G250, G350, G450 and G700 Media Gateways.

As with the S8400, S8500 and S8700 series Avaya Servers, the S8300 supports the Linux operating system and it is powered by Communication Manager. It supports industry standard call control, quality of service, management functions, and IP, digital, and analog endpoints.

Avaya S8300 Server

The S8300 is a media module that is inserted in any of Avaya’s G450, G700, G350 and G250 Media Gateways so size & weight will be different but for the proposed G350 the environmental information is listed below:




Avaya G450 Media Gateway Form Factor and Environmental Requirements

Physical Specifications

Dimensions Weight

Height 5.25 in (133.3 mm)

Width 19 in (482.6 mm)

Depth 18 in (456 mm)

Weight of empty chassis 7.4 kg (16.5 pounds) with blank plates 9.5 kg (21 pounds)

Power

100-240V~, 7 A Max, 50-60Hz

Environmental Conditions

Operating Condition Parameters

Temperature 0-40°C

Humidity 10-90%, non-condensing

Avaya G350 Media Gateway Form Factor and Environmental Requirements

Physical Specifications

Dimensions (H x W x D) Weight

133.3 mm x 482.6 mm x 400.0 mm 9-10 kg

Power

100-240V~, 7 A Max, 50-60Hz

Environmental Conditions

Operating Condition Parameters

Temperature +0 C - 40 C

Humidity 5%-90% relative humidity




1.3.1.2 Common Control Redundancy The IPTS common control must be based on a fully redundant duplicated design. Redundant components may be provisioned as active/passive or load sharing with seamless switchover operation between control elements in case of errors or failure. All active calls and programmed feature states must be preserved during the switchover process.

Vendor Response Requirement: Confirm that the proposed IPTS common control fully satisfies the requirement for a fully redundant duplicated design and identify if it is based on an active/passive or load sharing (pooled) design. In your response confirm which of the following elements are provisioned in fully redundant mode and specify the design option (active/passive, load sharing, pooled resources, et al.):

• Primary call processor

• Main system memory

• Customer database memory

• RJ-45 Ethernet uplinks to network

• Power supply

• Tone generators

• Call classifiers

• Registers

• DTMF receivers

• I/O interfaces

Avaya Response:

Avaya has proposed Simplex reliability for VoiceCon’s solution using LSP backup servers. The optional high reliability configuration could be achieved by adding a few optional pieces of equipment which consist of the following:

The high-reliability configuration option builds on the simplex-reliability option. The highreliability option duplicates components so that no single point of failure exists in the control network. The high reliability configuration consists of the following:

¾ Two Ethernet switches (Provided by VoiceCon’s network)

¾ Two UPS units (VoiceCon Provided)

The proposed solution has Local Survivable Processors running in each additional location CHQ2, RO, and SO.

The R0 and SO locations have an S8300LSP which includes Modem Dial Back-up. This feature provides a way to reconnect to the centrally located Avaya Server if the primary connection is lost by automatically dialing out over a modem to re-establish the connection.

The other site (CHQ2) is running an S8500 Server in LSP mode which can independently support up to 2400 IP terminals and 800 trunks. The survival ability will be designed by VoiceCon, with support form Avaya during installation, which best suits their needs.




1.3.1.3 Distributed Control Redundant duplicated common controllers must be installed in each of the two CHQ equipment rooms.

Vendor Response Requirement: Confirm that redundant duplicated common control elements are distributed and installed in each of the two CHQ equipment rooms. Describe how the distributed common control elements are physically and logically linked across the two equipment rooms.

Avaya Response:

Comply with explanation. The redundancy is via passive LSP S8500C servers (2) spread across CHQ as well as S8300 LSPs in both the RO and SO (up to 450 IP endpoints apiece). As stated earlier Avaya has proposed one S8500C Server that will control the whole VoiceCon Enterprise, meeting all capacities and growth requested. In addition each location can run independently, using each local LSP Server upon a total WAN failure.

The S8500C is connected via Processor Ethernet across VoiceCon’s LAN/WAN. If there was a failure each Media Gateway would re-register with any LSP in the Enterprise (one per location) based upon defined translations for example if CHQ2 lost its connection to the primary CHQ1 the first Server it might look for is the LSP at CHQ2 if that is not available then it might register to the LSP at CHQ1 or for certain Media Gateways try RO or SO locations (remembering the total capacity for the later is 450 IP endpoints.

1.3.1.4 Call Processing The proposed IPTS must handle a minimum of 150,000 Busy Hour Call Completions (BHCCS) in its proposed system design and fully equipped configuration.

Avaya Response:

Comply. S8500C Server at up to 60% occupancy can run 180,000 BHCC. The S8500C will support 180K calls per hour of IP endpoints, using Processor Ethernet (as designed for VoiceCon)

1.3.2 Call Processor Make/Model Vendor Response Requirement: Identify the make/model of the main processor for the proposed common control system.

Avaya Response:

The proposed system uses the Avaya S8500C Server (IBM Model X306M) with a 3.2GHz 2216 Processor and two 512MG of memory




1.3.3 Call Processing O/S Vendor Response Requirement: Identify the primary operating system used by the main processor of the proposed common control system. Linux is preferred, but not mandatory.

Avaya Response:

The proposed Avaya S8500C Server uses the Enterprise Linux 4.0 Operating System.

1.3.4 Memory Vendor Response Requirement:

Briefly describe system memory and storage design for both generic software and customer database requirements.

Avaya Response:

Comply. The memory specifications for the Avaya S8500 Server are as follows:

¾ 2 - 512MG PC2-4200 CL4 ECC DDR2 SDRAM DIMM

¾ Optical Drives: DVD/CD-ROM

¾ NIC:

� 2 NICs (10/100/1000)

� 4 NICs (10/100)

¾ Hard Drive: 1 80GB SATA HDD drive.

¾ Ports: 3 USB Ports

◊ 1 Compact Flash (external USB)

◊ 3 USB ports unused

1.3.4.1 Database Integrity Vendor Response Requirement:

How does the proposed IPTS solution maintain the integrity of the customer database between backups?

Avaya Response:

To protect the integrity of the data, the "save translation" command writes two time-stamped identical copies of the configuration data to the selected memory card, disk, or tape. The save writes one complete copy first, and then writes the second copy in a different area of the device — both with the same time-stamp. Failure during a save, including a system crash, usually affects only one copy. The affected copy is marked "bad" and should not be used for backup.

Best practice would recommend off-site storage for all valuable company databases, including the telephony server database. In addition to daily maintenance, the Web Services inherent in any Avaya Linux-based Server can FTP (or email) the translation database to an off-site server (preferably at a hardened location with RAID drives) on a daily basis.




1.3.4.2 Database Information Loss Vendor Response Requirement:

Identify under what circumstances can customer database information (configuration, messages, logs, etc.) be lost during back-ups

Avaya Response:

The standard daily maintenance cycle would save database changes / translations to an Avaya Linux-based Server hard drive, and to any Enterprise Survivable Servers (ESS) (not included in this proposal) and Local Survivable Processors (LSP) hard drives in your network. Therefore move, add, and change activities completed for a particular day are saved to the Avaya Server’s hard drive and distributed to ESS and LSP hard drives automatically.

Any changes made between daily maintenance cycles would be lost unless the administrator uses the “save translation” command.

1.3.4.3 Database Backup Scheduling Vendor Response Requirement:

How often should the customer database be backed up? Specify if it is a full or incremental backup and the time the process takes.

Avaya Response:

One of the most important administrative tasks is to set up a schedule to routinely back up system data to a safe location. While the detailed procedure is up to VoiceCon, Avaya supports multiple backup options. A backup procedure can run manually or automatically based on a schedule created by your administrator. Possible destinations for the backup files are an FTP server, email, and local PC card. The web interface allows the administrator to back up call-processing data (Avaya Communication Manager “translations”), server system data, and security files. Avaya recommends that the system administrator encrypt the backup files to keep this sensitive information secure.

Administrators can use a browser with SSH to access the server and schedule backups, as well as restore the configurations. Alternately, a locally connected console can be used to store/restore the configurations.

We recommend that backups be done at least once per day (especially of the translation set). The backups are “full”; i.e. they contain all of the data specified in the set, not just the modified files. Typically, a backup set for Communication Manager will take on average about 3 minutes or less. However, this is an average value, because the “translation” set can take longer dependent on the size of the translation file, and if a save translation has been requested.




1.3.4.4 Data Purging/Archiving Vendor Response Requirement:

Describe the mechanism for data purging and archival, including storage and retrieval of archived data.

Avaya Response:

The storage of the images is VoiceCon’s responsibility. That is we provide the mechanism to create the images; but VoiceCon determines where the images are to be retained (remote server via ftp, etc, or compact flash). We do not support or recommend storing the images to the local server, for the mere fact that if the servers become inaccessible, VoiceCon will no longer be able to access his archives.

The user has the ability to see the archive information for the machine via the web menus or they can issue a query directly to the destination device/system; in this particular instance, the query will display a list of archival files that are at the specified destination.

None of the images are ever overwritten by Avaya. Each image has a unique date/time stamp as a part of the image name; therefore the files will remain at the destination server until removed by VoiceCon. In the case of Compact Flash destinations, the user has the option to specify how many copies of each archive type they want to be kept on the system (up to a maximum of 9 copies, It will always be the latest 9 backup images).

1.3.5 Power Supply Vendor Response Requirement: Briefly describe common control power requirements and the integrated power distribution design. Indicate if the power supply is dependent on either an AC or DC current source.

Avaya Response:

Each S8500C Server uses an auto-ranging 100-240 Vac power supply, 50 to 60 Hz, 5 A maximum at 100-120 Vac and 2 A maximum at 200-240 Vac. The AC power source is to be single phase, 3-conductor (Line, Neutral and Ground) with a 15A circuit breaker for 100-120 Vac or a 10A circuit breaker for 200-240 Vac.

1.3.5.1 Power Safeguards

Vendor Response Requirement: Describe any power failure safeguards that are included in the IPTS design. Briefly describe what happens to system operation during a power failure

Avaya Response:

If the active Avaya Server has a complete catastrophic failure a Local Survivable Processor (LSP) Media Server takes over. During a complete power failure at the CHQ1 site the power failure transfer equipment will activate allowing access to local two-way CO trunks via VoiceCon provided analog telephones. At that point, the other sites will no longer be able to “communicate” and will activate its LSP mode allowing any IP telephones or “powered up” media gateways to register as defined in the translations.




IP Telephones and media gateways receive a list of alternate gatekeepers as part of the standard configuration. If the endpoints cannot reach their primary gatekeeper, the endpoints will re-register with the secondary gatekeeper. If they cannot reach that gatekeeper, they will attempt to register with the tertiary gatekeeper (an LSP at the Small Branches), and so on.

Business Continuity

G250/G350/G450/G700 media gateways provide a very compelling offering for the growing distributed branch office market. LSP capacity now matches H.248 media gateway capacity, which means branch office deployments, can now be fully deployed with LSP support. To support this increased capacity, a new LSP translation synchronization feature improves bandwidth utilization for file updates.

Enterprise Survivable Server (Optional)

The Optional Enterprise Survivable Servers, not part of this solution, allow backup servers to be placed at various places in a network so that communications can continue in the event that a main server(s) fails, or when connectivity to the main server(s) is lost. Enterprise Survivable Servers can be connected to ATM, IP and Center Stage Switch (CSS) connected port networks. The IPSI card in the port networks automatically obtains service from an ESS server(s) if the control signal to the main server is lost. In Avaya Communication Manager 5.0, there can be up to 63 Enterprise Wide ESS servers in a network.

Deploying Enterprise Survivable Servers in addition to LSP as requested at appropriate points in a network provides flexibility in how to plan against system failover. Assurance of communications continuity can be mapped to protect against network failures or catastrophic main server failures or both. Administration for the ESS can be conducted in a central point with automatic synchronization to all ESS servers.

1.3.5.2 Power Backup

Vendor Response Requirement: Does the proposed IPTS solution come equipped with standard UPS hardware, and if so how long can the system run on it? If not, what UPS requirements are recommended?

Avaya Response:

The Avaya solution does not include UPS Hardware. The Avaya Servers are always recommended with power backup per server. Power backup is required to avoid power problems and to ensure graceful shutdown of the system processes if the power fails. Combinations of battery extension modules and a 1500-VA UPS can provide up to eight hours of power backup.


individual consumption.



Avaya considers voice communications to be mission-critical to customers' enterprises, and strongly recommends the use of an online Uninterruptible Power Supply (UPS) with each S8X00 Server. VoiceCon must provide a UPS to support each Server with a 400VA and 3.33AMP rating for each server. Avaya has priced UPS as an option for this solution.

1.3.5.3 Power Consumption1

VoiceCon desires an IPTS with reduced power consumption requirements compared to earlier generation system designs.

Vendor Response Requirement: Provide data regarding common control power consumption requirements and heat dissipation levels (BTUs) for the proposed IPTS solution.

Avaya Response:

The Heat dissipation for an S8500 Server is:

¾ Minimum configuration: 341 BTU/H (100 W)

¾ Maximum configuration: 1024 BTU/H (300 W)

A fully loaded G450 produces 1364 BTU/H

1.3.5.4 Environmental Initiatives

Vendor Response Requirement: Briefly identify any current or planned IPTS design initiatives to reduce power consumption requirements and heat dissipation levels.

Avaya Response:

Avaya and its strategic global partner Extreme Networks® have demonstrated how an Engaged Network for Intelligent Communications can deliver real energy cost savings while offering a converged network for IP telephony at Gigabit speeds during the VoiceCon 07 fall show. The “Go Green at Gigabit Speeds” initiative was discussed during a presentation outlining how the combination of Avaya’s IP Communication Solutions and Gigabit IP phones with Extreme Networks Universal Port software can automatically help optimize the power consumption of connected devices, thus reducing electricity use. It provides precise control

over individual workstations and desk phones and other Power of Ethernet (PoE)-capable devices, enabling you to maximize power efficiency

Helping to save the planet means everyone must do his or her part – at home and at the office. And Avaya and Extreme Networks are doing their part by providing products now which reduce electrical consumption –a win-win-win scenario!

How it works

With Avaya’s power-efficient IP desk phones and the automation of the Universal Port™ feature contained in Extreme Networks ExtremeXOS™

network operating system, you get an IP desk phone solution that uses less power to begin with and gives administrators the ability to manage

or group-based resources easily, for more efficient electrical

1 1.3.5.3 & 1.3.5.4 added buy Allan 11/5/07




Take, for example, an office-based sales force that works a regular 9 to 5 shift. In most current scenarios, the sales force’s phones are powered on around the clock, 24x7, consuming wattage unnecessarily. But with the combined GigE solution, those phones – energy efficient to begin with – become even less of a drain on power, as their power profile is controlled by the Extreme Networks’ intelligent Universal Port capability. Phones can easily and automatically be powered down at 5:00 in the evening and then restarted at 9:00 the next morning (while still allowing for individual exceptions for late workers and early risers). If you also power down on weekends, that’s an immediate 75% potential savings in electrical consumption for the block of phones under automated control – a significant reduction.

¾ Avaya IP Desk phones are designed to perform using the least amount of power possible. The state-of the- art 9630G and 9640G IP Desk phones – with native GigE compatibility – are rated Class 2 PoE. Comparable native GigE phones from Cisco, Nortel and ShoreTel are rated at a less efficient Class 3 PoE.

Other GigE-capable IP phones from Avaya, including others in the 9600 Desk phone series as well as the 1600 and 4600 Desk phone series, are also built with efficiency in mind. These can be adapted to the gigabit Ethernet using an available Avaya Gigabit Ethernet Adapter.

¾ Extreme Networks Universal Port gives network administrators and facilities managers the ability to save electricity by automating the PoE port configurations for individual or blocks of IP phones not in use for substantial periods of time.

Extreme Networks unique Universal Port feature is a powerful framework for time or eventdriven activation of automated profiles that in addition to PoE control can also support automated phone provisioning and authentication to simplify the management of the network infrastructure.

As more businesses move to GigE – rapidly being adopted as the new standard in enterprise networking – managers are looking for economical, sustainable solutions that leverage the benefits of the technology while lessening its costs. The combined GigE solution from Avaya and Extreme Networks allows companies to be on the leading edge of technology, exercise fiscal prudence and reduce unnecessary electricity consumption to help save the environment.




1.3.6 Ethernet Call Control Signaling Links Vendor Response Requirement: Identify for each active and standby call telephony server the number of available and configured RJ-45 Ethernet LAN uplink interfaces for call control signaling to LAN-connected cabinets/carriers and/or standalone ports. Include a brief description of how the physical Ethernet connection is provided: dedicated circuit board; daughterboard; fully integrated RJ-45 connector, et al.

Avaya Response:

The Avaya Media Servers provide integrated RJ-45 connectors. A connection is established between the Avaya Servers and the customer-provided layer-2 devices. From the data switches, a connection is established to a dedicated circuit pack.

In the case of the Avaya Media Gateways, control signaling begins at the Avaya Servers, traverses the control network through the Control LAN (C-LAN), terminating at the G450 and G350 Media Gateway on a integrated RJ-45 port. VoiceCon’s data infrastructure provides the transport between Server and G450 and G350 Media Gateway, depending on the configuration.

As with the G450 and G250 Media Gateways, all IP endpoints terminate signaling traffic on the C-LAN gatekeeper.




The front panel of the Avaya S8300 Server

The S8300 Server is a Pentium-based processor that runs on a Linux operating system. The S8300 runs Avaya Communication Manager to provide call control services to the G350 and other Avaya gateway devices.

The front panel of the S8300 includes:

¾ 10/100BaseT Fast Ethernet port (SERVICES)

¾ Two USB ports for modem connections or the USB CD-ROM drive (USB 1 and USB 2)

The S8300 Server




1.3.7 System Clocks Vendor Response Requirement:

Identify the number and type of internal system clocks that are available and configured.

Avaya Response:

The Avaya G350 and G450 Media Gateway use a VoIP Motherboard supported Clock Generation as well as synchronization

The G350 and G450 also support the following capabilities:

¾ 15 channels of tone detection in each G350, 64 channels in the G450

¾ Tone generation

¾ 15 minutes of announcement storage in the G350, 45 minutes on the G450

¾ 6 announcement playback channels in the G350, 63 announcements in the G450

¾ LAPD termination

¾ H.248 signaling to the Controller

¾ Gateway maintenance

The G450 will support 63 announcement ports for announcement play back. The G450 also provides 1 port for announcement recording. In R1, the G450, in its base unit configuration, the onboard flash stores up to 30 Minutes of announcements and 15 minutes of MOH (Music on Hold), in G711 quality, all totals to 45 minutes storage and all 45 minutes can be used for announcements and/or MoH.

1.4 Remote Survivability In standard operating mode station users at RO and SO facilities will be supported by CHQ common control. It is required that station users at the RO and SO facilities be provided with full, uninterrupted access to all IPTS features and services regardless of CHQ common control failure or LAN/WAN connectivity issues due to switch, router, or private network transmission service errors or failures. Each remote facility must perform a seamless switchover to a local secondary common control while all active local intercom and trunk calls, and programmed feature states, e.g., call forwarding, are preserved.


Confirm that the proposed IPTS solution satisfies this requirement for remote local survivability and briefly describe the local hardware/software solution at each of the two remote facilities necessary to support the seamless switchover operation. Indicate the time it takes to perform the switchover if not instantaneous and what happens during this time period, including access and implementation of generic software features and functions.2

2 Red was added via revised RFP document 11/5/2007




Avaya Response:

The Avaya Local Survivable Processor (LSP) solution utilizes the Avaya S8500 and/or S8300 Server hardware and software components and requires an Avaya MultiVantage Software license to activate the LSP feature. This software license allows the Server to be a survivable call-processing server for remote/branch customer locations, as well as a redundant call controller for a standalone S8300 Server. In the event that the communication link is broken between the remote Avaya G450 or G350 Media Gateway and the primary call controller (either an Avaya S8300 Server or an Avaya S8500 Server), the LSP will provide service for the Avaya IP Telephones and Media Gateways that were controlled by the primary call controller.

The strategy by which Avaya Gateways and IP endpoints change control from the primary gatekeeper to the LSP is driven by the endpoints themselves, which use a list of gatekeepers. During initialization, each IP endpoint and Media Gateway receives a list of gatekeepers. The IP endpoints ask each gatekeeper in the list for service until one responds with a positive reply. If the link to that gatekeeper fails at some later time, the endpoint will try to receive service from the other gatekeepers in the list, including the LSP. The LSP will provide service to all IP endpoints that register with it.

When the primary gatekeeper is prepared to provide service, the LSP needs to be manually reset. This will inform the IP endpoints to try their gatekeeper list again, and return to the primary gatekeeper for service. The LSP provides redundancy in a variety of configurations and can be located anywhere in a network of Media Gateways.

Using the defaults of one-minute primary search time, a transition point of one, a total search time of 30 minutes, and one LSP with a network failure, the following are the times involved in the Avaya G700 Gateway failover to an LSP:

¾ Keep-alive failure : 72 seconds

¾ (first) Search time attempt to the CLAN : 10 seconds

¾ Connect to LSP : ~1 second (can be up to 10 seconds)

¾ Media Gateway Processor (MGP) reboot : 40 seconds

¾ Search time attempt to the CLAN: 10 seconds

¾ Reconnect to LSP : ~1 second (can be up to 10 seconds)

Until the Avaya Gateway reaches a second gatekeeper, all calls in progress will remain up, but without access to features. Any call that is attempted when the gateway is in its recovery stage will receive no dial tone until the Media Gateway re-registers with a gatekeeper and is receiving call processing again. After the Gateway reaches a second gatekeeper, the Media Gateway Processor resets and any DCP, analog, or trunked calls in progress at this time are torn down.




Any IP-direct calls (In network - IP endpoint to IP endpoint) will stay up until both parties hang up, but will have no access to any features until a new call is originated. The telephone will enter into recovery mode when the users hang up. The LSP will deny IP phone registrations until at least one Media Gateway has registered with the LSP. Until the Gateway has registered, the LSP has no access to IP resources to serve the telephones, so this is a precautionary measure to ensure the highest quality of service once the transition to the LSP has occurred.

Once the primary call controller is ready to serve the endpoints being served by the LSP, the LSP must undergo a manual reset, and the endpoints will re-register with the primary call controller once again.

Any LSP that controls an Avaya Gateway with a trunk, either to the PSTN or to other call controllers, will have access to that trunk. This is important because once LSPs are giving service to Gateways and IP telephones; each LSP is its own switch and has no communication paths to other LSPs. Thus, a DCP phone on LSP1 can call a DCP phone on LSP2 through PSTN trunks maintained in each LSP. The same rule applies for IP telephones registered to different LSPs; no direct extension calls between LSPs can be placed.

Traditionally, duplication of data has been done by a call processor, which protects against server difficulties and provides this protection invisibly. When increasing the distances between headquarters and remote branch offices however, it becomes more likely that a network outage causes a loss of communications. Setting up Local Survivable Processors on corporate networks provides coverage with full feature functionality in the event of a network outage. With the Avaya MultiVantage Software based solutions, the system administrator is empowered to select which strategy for recovery is priority: reconnecting the location to the network for interoffice communications, or reestablishing local call traffic.

1.4.1 Survivable IPTS Features/Services Vendor Response Requirements:

Identify any required generic software feature or (See Section 5.0 Call Processing Features) or system function (including E911 support services) not available or operational when the local survivability option is activated at either the RO or SO facility. Also identify any type of station user equipment (instruments, consoles, soft phones, mobile clients, et al) not supported in local survivability mode at either of the two facilities.

Avaya Response:

Comply. All features and capabilities supported by Avaya Communication Manager are available to users at each site when in survivable mode.

The proposed solution includes an S8500C Server at the CHQ1 location and S8500C LSP servers at the CHQ1 and CHQ2 as well as the Regional Office (RO). The in the event the WAN is lost to CHQ1 and there is a connection available to either the CHQ2 or RO all offices would re-register to the defined facility and continue operations as normal.




The SO will have an S8300 Local Survivable Processor (LSP) The LSP, an S8300 Server, is a blade housed within the Avaya H.248 Gateway at each location. The LSP maintains a copy of the S8500 Server customer translations. An automatic process copies customer translation changes made on the primary server to every LSP in the network. The translations are updated regularly using a virtual link via the IP network. To maximize effectiveness while conserving bandwidth, a file difference mechanism identifies changed records within the translation file and transmits only changed records when file synchronization occurs.

S8300B Media Server Module

Typically, all LSPs are in idle mode, where the LSP is not processing any calls. When the Media Gateway’s Processor (MGP) or IP endpoints perceive the Avaya Server to be unreachable, the MGP or IP endpoints will attempt to register with a predefined LSP. The LSP does not actively take over when the primary controller becomes unreachable, but waits for MGPs and IP endpoints to register with it. If for any reason communication between an Avaya H.248 Media Gateway and its primary controller stops, the LSP activates. This "fail-over" from the primary controller to the LSP is an automatic process without human intervention. The LSP assumes control of any IP telephone provided that telephone has the LSP in its list of controllers. A feature called Connection-Preserving Failover/Failback for H.248 Media Gateways provides uninterrupted communication service for stable calls at the branch office when switching from an active/main component to a standby/backup component or vice versa.

IP telephones obtain their own IP address from a DHCP server. The DHCP server also sends a list of controllers, LSPs, and their associated IP addresses. The IP telephone then registers with the controller corresponding to the first IP address in this list. When connectivity is lost between the controller and the endpoint, the endpoint registers with the second IP address in the list, and so on. This list is administrable for telephones on the DHCP server.

For each location, PSTN trunks were requested per your specifications, so the requested PSTN facilities will support the location in case of a WAN failure.

The Connection Preserving Migration (CPM) feature preserves existing bearer (voice) connections while an H.248 media gateway migrates from one Communication Manager Media Server to another. Migration might be caused by a network or server failure. Only stable calls are preserved. Calls that are not preserved during the transition to LSP mode are:

¾ Unstable calls. An unstable is any call where the call talk path between parties has not been established, or is not currently established. Some examples are:

¾ Calls with dial tone

¾ Calls in dialing stage

¾ Calls in ringing stage

¾ Calls listening to announcements

¾ Calls listening to music

¾ Calls on hold (soft, hard)




¾ Calls in ACD queues

¾ Calls in vector processing

¾ IP trunks, both SIP and H.323

¾ ISDN-BRI telephones

¾ ISDN-BRI trunks

Users on connection-preserved calls cannot use such features as Hold, Conference, or Transfer.

1.4.2 Local Survivability Failover and Switchback Vendor Response Requirements:

For each of the remote facilities confirm that switchover to the local survivable common control option seamless, i.e. no interruption of in-process telephony services, for any or all stations users if WAN connectivity is disrupted to the HQ IPTS? Also provide answers to the following:

• Is there a station user recognizable delay in dial tone and call implementation after the WAN disruption?

Avaya Response:

Until the Avaya Gateway reaches a second gatekeeper, all calls in progress will remain up, but without access to features. Any call that is attempted when the gateway is in its recovery stage will receive no dial tone until the Media Gateway re-registers with a gatekeeper and is receiving call processing again.

• Describe the switchback process when HQ facility IPTS common control is again accessible, specifying if the process is automatic or manual and how long the process takes to implement

Avaya Response:

A feature of Avaya Communication Manager called Auto Fallback to Primary for H.248 Media Gateways automatically returns a fragmented network, where a number of H.248 media gateways are being serviced by one or more Local Survivable Processors (LSP), to the primary Server.

Auto fallback to primary for H.248 Media Gateways allows your administrator to define any of the following rules for migration/fallback to the main server:

¾ Whether or not the media gateways, serviced by ESS/LSPs, shouldautomatically migrate to the primary media gateway.

¾ Whether or not the media gateway should migrate immediately whenpossible, regardless of active call count.

¾ Whether or not the media gateway should only migrate if the active call count is 0.

¾ Whether or not the media gateway should only be allowed to migratewithin a window of opportunity, by providing day of the week and time intervals per day. This option does not take call count into consideration.




¾ Whether or not the media gateway should be migrated within a window of opportunity by providing day of the week and time of day, or immediately if the call count reaches 0. Both rules are active at the same time.

Internally, the primary call controller gives priority to registration requests from those media gateways that are currently not being serviced by an LSP

• Are connected calls and/or telephony server operations at the remote facility affected in any way during the switchback process? If the answer is yes, describe how they are affected.

Avaya Response:

Yes, since the return to the primary call controller is not call preserving, it is recommended that this outage be planned and only done when service will be the least affected.

The return of service to the primary call controller requires manual intervention. The LSP must undergo a reset system 4, which will close all communication links with the LSP, forcing all endpoints to reregister with the primary call controller. This time to transition is again dependent upon the ordering and administration of alternate controller lists, but the initial discovery that the connection has been broken is immediate. Both IP endpoints and Gateways will discover right away that the link is no longer usable and begin the search for another controller.

1.4.3 Survivable Messaging Services It is desirable that remote station users at the RO and SB facilities have access to messaging services (see Section 7) if there is a WAN link disruption to the HQ messaging system.

Vendor Response Requirements

Briefly describe how messaging services would be accessed and implemented by remote station users when there is a WAN link disruption. Would any messaging features/functions not be supported in this situation.

Avaya Response:

The proposed messaging solution is centralized at the CHQ location with access to the

can still leave messages at the central messaging system. End user’s can access their messages via the TUI by dialing through the PSTN. This would preclude them receiving message waiting and any GUI access to messages. The minimum messaging requirements are met in the recommended configuration.

1.4.4 Network Failover Resiliency VoiceCon desires that the proposed IPTS support network failover resiliency to a comparable IPTS located at a remote facility in the unlikely

carrier/media gateway equipment at all facilities (CHQ, SO, and BO) can automatically re-register to a designated back-up IPTS if so programmed.3


Respond to each of the following:

3 Revised 11/5/07 by Allan

branch location(s) via the IP connections. In the event the IP Connection (WAN) fails, callers

event the redundant CHQ common control (primary active/active or active/passive) becomes nonfunctional for any reason? Network failover resiliency requires that all IPTS station users and port




• Can the proposed IPTS solution support network failover resiliency to another back-up IPTS in case of a catastrophic CHQ common control failure?

• If yes, briefly describe the failover process including the time required before full telephony services are available to re-registered station users.

• Can there be more than one designated back-up IPTS?

Avaya Response:

Comply. The two S8500 servers running in Local Survivable Processor mode (LSP) positioned at CHQ1 or CHQ2 can support all licensed ports individually in the event the active S8500C at CHQ1 fails.

IP Telephones and media gateways receive a list of alternate gatekeepers as part of the standard configuration. If the endpoints cannot reach their primary gatekeeper, the endpoints will re-register with the secondary gatekeeper. If they cannot reach that gatekeeper, they will attempt to register with the tertiary gatekeeper, and so on. VoiceCon will be able to decide the order that suits their needs.

Using the defaults of one-minute primary search time, a transition point of one, a total search time of 30 minutes, and one LSP with a network failure, here are the times involved in the Avaya G700 Gateway failover to an LSP:

¾ Keep-alive failure : 72 seconds

¾ (first) Search time attempt to the CLAN : 10 seconds

¾ Connect to LSP : ~1 second (can be up to 10 seconds)

¾ MGP reboot : 40 seconds

¾ (second) Search time attempt to the CLAN: 10 seconds

¾ Reconnect to LSP : ~1 second (can be up to 10 seconds)

1.5 Session Initiated Protocol (SIP) VoiceCon requires that the proposed IPTS support SIP-compatible stations and trunk networking as specified by IETF Work Group RFC documents, including 3261, 3263, 3264, 3265, 3604 and 4456,.


Respond to each of the following:

• Does the proposed IPTS conform to IETF SIP specifications cited above 3261?

• Is the proposed IPTS solution based on a native-SIP design or is optional hardware/software required? Indicate if optional equipment is required to support SIP communications protocol specifications.

• Can the proposed IPTS support SIP-compliant desktop telephone instruments?

• Can the proposed IPTS support SIP trunk services? Indicate if optional SIP proxy gateways are required?

• Can the proposed IPTS support SIP-enabled applications, such as Internet conferencing, telephony services and features, presence, events notification and instant messaging? Indicate if optional server equipment is required.




Avaya Response:

Comply.

However RFC 3604 & 4456 have nothing to do with SIP! Please see below links for what they address.

Requirements for Adding Optical Support to the General Switch Management Protocol version 3 (GSMPv3) http://www.ietf.org/rfc/rfc3604.txt

BGP Route Reflection: An Alternative to Full Mesh Internal BGP (IBGP) http://www.ietf.org/rfc/rfc4456.txt

Avaya offers SIP endpoints and SIP trunking. SIP is supported natively on Avaya Communication Manager and SIP trunking can be supported directly or via the optional SIP Enabled Services (SES), which provides additional services including SIP Proxy, SIP Registrar, SIP Location Server, and SIP Presence/Event Server. SIP endpoints require the SES server and all SIP telephony users are administered in and register to the SES. Adding an Avaya SES to your enterprise enables the following:

¾ Avaya Communication Manager becomes a telephony feature server accessible from any SIP endpoint – allowing access to the rich Avaya Communication Manager feature set to support capabilities not currently provided by SIP

¾ Enterprise Instant Messaging (IM) functionality – a component of Avaya IP Softphone Release 5.x or greater and IP Agent Release 6.x or greater. Think of this as an advanced IP Softphone application with an IM client that integrates a buddy list of voice and IM communication. IM messages are sent encrypted and can be archived to meet corporate communications standards

¾ Our SIP telephones offer the standard telephony features of SIP and are more tightly integrated with the value added features of Communication Manager. These are the same IP telephones that are proposed, but with a different load of firmware (SIP vs. H.323 load)

¾ SIP trunking which leverage service provider media gateways

¾ SIP integrates the communication silos of your voice portal (office phone and cell phone) and text messages (IM and email) into one logical address to reach a user

¾ Improves productivity by reducing phone tag with "presence awareness"

¾ Interoperable products from multiple vendors which offer additional choices

¾ The Avaya implementation allows our customers to implement SIP functionality while still providing the robust features of Communication Manager that the SIP standard does not address.

The Avaya implementation will allow any Avaya Communication Manager endpoint (DCP, analog, IP) to be addressable with a SIP URL to protect your initial investment

Avaya SIP Enablement Services (SES)

Avaya provides an optional standards-based SIP architecture for telephony, Instant Messaging, and other enterprise communications. SES extends the power of Avaya Communication Manager to SIP telephony endpoints, and works with Avaya IP Softphone and one-X Desktop edition SIP hard phones to introduce presence-based telephony. The SES runs on top of Linux on one or more Avaya Media Server platforms, enabling you to integrate SIP-based and conventional telephony.




SIP Enablement Services Server

SIP Enablement Services 4.0 and later supports the following SIP-related IETF RFC's and Internet Drafts:

¾ RFC 2246 The TLS Protocol

¾ RFC 2396 URI generic syntax

¾ RFC 2617 Digest Authentication

¾ RFC 2782 A DNS RR for specifying the location of services (DNS SRV

¾ RFC 3261 SIP: Session Initiation Protocol

¾ RFC 3262 Reliability of Provisional Responses in the Session Initiation Protocol (SIP)

¾ RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers

¾ RFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP)

¾ RFC 3265 SIP-Specific Event Notification

¾ RFC 3311 UPDATE method

¾ RFC 3325 Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks

¾ RFC 3428 Session Initiation Protocol (SIP) Extension for Instant Messaging

¾ RFC 3515 REFER method

¾ RFC 3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)

¾ RFC 3856 A Presence Event Package for the Session Initiation Protocol (SIP)

¾ RFC 3858 XML Format for Watcher Info

¾ RFC 3911 The SIPHeader

¾ RFC 4028 Session Timers in SIP

¾ RFC 4244 Request History Information

¾ draft-elwell-sipping-redirection-reason-00

¾ draft-ietf-impp-cpim-03

¾ draft-ietf-impp-cpim-pidf-07

¾ draft-ietf-simple-presencelist-package-00

¾ draft-ietf-sipping-cc-conferencing-04

¾ draft-ietf-sipping-cc-transfer-02

¾ draft-ietf-sipping-conference-package-01

¾ draft-ietf-sipping-dialog-package-04




SIPConnect is an industry standards-based approach to direct IP peering between SIPenabled IP PBXs and VoIP service provider networks. The SIP Forum began working onSIPConnect subsequent to the initial release of the SIPConnect Interface Specification inFebruary of 2005 based on a proposal submitted by members of the SIPConnect initiative.Avaya is a founding member of the SIPConnect initiative.

Visit the SIPConnect technical working group at www.sipforum.org for further details or todownload the technical recommendation.

Communication Manager 4.0 and later supports the following SIP-related IETF RFC's and Internet Drafts:

¾ RFC 1889 RTP: Real-Time Transport Protocol

¾ RFC 2246 The TLS Protocol

¾ RFC 2327 SDP: Session Description Protocol

¾ RFC 2396 URI generic syntax

¾ RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

¾ RFC 3261 SIP: Session Initiation Protocol

¾ RFC 3262 Reliability of Provisional Responses in the Session Initiation Protocol (SIP)

¾ RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers

¾ RFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP)

¾ RFC 3265 SIP-Specific Event Notification

¾ RFC 3311 UPDATE method

¾ RFC 3325 Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks

¾ RFC 3515 REFER method

¾ RFC 3551 RTP Profile for Audio and Video Conferences with Minimal Control

¾ RFC 3578 Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signaling to the Session Initiation Protocol (SIP)

¾ RFC 3711 Secure Real-time Transport Protocol (SRTP)

¾ RFC 3840 Indicating User Agent Capabilities in the Session Initiation Protocol (SIP) (partial support)

¾ RFC 3841 Caller Preferences for the Session Initiation Protocol (SIP)

¾ RFC 3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)

¾ RFC 3891 The Session Initiation Protocol (SIP) "Replaces" Header

¾ RFC 3911 The SIP Header

¾ RFC 3960 Early Media

¾ RFC 4028 Session Timers in SIP

¾ RFC 4244 Request History Information


products



¾ RFC 4568 Security Descriptions for Media Streams

¾ draft-elwell-sipping-redirection-reason-00

¾ draft-ietf-sipping-cc-conferencing-04

¾ draft-ietf-sipping-cc-transfer-02

¾ draft-ietf-sipping-conference-package-01

¾ draft-ietf-sipping-dialog-package-04

¾ draft-ietf-sipping-realtimefax-01

1.6 Security VoiceCon requires a secure IPTS network solution to optimize system performance and reduce the probability of toll fraud, restricted calls, and illegal system and network access.

1.6.1 Unauthorized System Access VoiceCon requires that the proposed IPTS solution be secure against unauthorized system access. The following system design and configuration guidelines should be followed:

• All unnecessary ports, such as telnet, SNMP, etc. will be closed by default.

• The software running for ports will not contain any known vulnerabilities.

• Administrative interfaces will not ship with known default passwords.

• Default community strings for SNMP will not be used. SNMP version 3 will be supported.

• The switch network will support security features such as VLANs, Network Admission Control (NAC), and other features.

• Key components, such as the call processor, media gateway, or associated servers/cards will have built in host-based intrusion prevention systems.


Confirm that the proposed IPTS solution satisfies each of the above listed security attributes. Briefly describe authentication processes embedded in the proposed IPTS solution to prevent unauthorized access to common control elements, data resources; and abuse of telephony services, e.g., toll fraud.

Avaya Response:

The Avaya Servers run a hardened Linux distribution with all unnecessary/unused services and packages removed or disabled by default. Viruses and worms are most commonly

delivered via e-mail, by visiting infected web sites, or by sharing of disk drives. The Avaya S8X00 Servers do not support incoming e-mail, does not support forwarding of E-mail (since there is no incoming e-mail in

the first place), do not contain a web browser, and does not support NFS or SMB (i.e. does not share drives). All file transfers to the Avaya

S8X00 Servers are restricted and are cryptographically signed to prevent introduction of unwanted (Trojan) software.

Avaya actively monitors security notification lists for vendors of which are incorporated into Avaya products in addition to

government agencies which report on vendor-agnostic vulnerabilities. This information is used to keep track of security vulnerabilities which may impact Avaya products and is also incorporated into product development processes to reduce the changes of shipping software with known vulnerabilities.




All software has vulnerabilities which may be discovered after product shipment so additional processes are in place to effectively deal with vulnerabilities discovered after the product ships (reference: 1.6.9 Miscellaneous Security Requirements).

Administrative interfaces do not ship with default passwords. Avaya maintenance access to the server requires multiple levels of strong (two factor) authentication using a one-time password challenge/response system.

SNMP is disabled by default and there are no default read, read/write community strings. SNMPv3 is supported.

The Avaya S8X00 Server development team investigated (and continues to investigate as this is not a static area) adding virus scanning software as part of the software operating on the server. Virus scanning software for Linux is not common, generally because the market and threats are not sufficiently large to entice scanner vendors. In addition, those vendors that do offer Linux scanners primarily scan E-mail (Avaya S8X00 Media Servers have no incoming/SMTP server so there is nothing for these products to scan) to prevent the spread of a virus to Windows based machines on the network; they do little specifically for the Linux box itself. A final and no less important consideration is the impact the introduction of additional software has on the reliability and overall stability of the platform. The more software/processes introduced, the greater the potential impact on reliability and stability. All of these different areas were weighed against the overall risk of malware being introduced into the system. At this time, the risk of malware introduction does not merit the inclusion of virus scanning software on the server.

Avaya Communication Manager utilizes the Tripwire host based intrusion detection system to provide notification of unauthorized or suspicious changes in key files and directories while the media gateways and associated LAN interfacing cards can notify based on usual traffic which may be indicative of a denial of service attack. Support for external system logging allows events to be aggregated into an enterprise monitoring system.

All access to privileged services require authentication. Once successfully authenticated, either for telephony administration or end-user services, administered authorization restrictions and defined auditing rules mitigate the risk of abuse of telephony services.




1.6.2 Unauthorized Network Access VoiceCon requires that the proposed IPTS solution be secure against unauthorized network access.


Briefly describe call type detection and prevention processes embedded in the proposed IPTS solution to identify and prevent:

• Unmonitored and non-secured Internet sessions by employees calling private Internet Service Providers accounts using modems connected to corporate phone lines.

• Unlawful data network access by outsiders penetrating through modemenabled corporate phone lines connected to LAN/WAN accessible workstations and other equipment.

Avaya Response:

Comply. Typical mechanisms of server access include Telnet, Web Browser (HTTP), and FTP for file transfer. Each of these mechanisms can support login authentication, but suffer a common weakness. During the login sequence, the username and password being supplied by the user is sent in clear text. This allows a person with a network monitor/sniffer to capture the password and gain access. In addition these mechanisms transmit all the session information in clear text. Some of this information might contain data such as account codes, authorization codes or other data useful to an attacker. To overcome these problems, Avaya S8X00 Servers support Secure Shell Access (SSH), Secure Copy (SCP), and Secure File Transfer Protocol (SFTP) and secure Web access (HTTPS).

¾ SSH, SCP, SFTP provide an access mechanism for terminal access and file copy that encrypt the entire session including the login sequence as well as subsequent data transfer.

¾ HTTPS provides a similar mechanism for Web access. All Web access to an Avaya S8X00 Server is via a secure connection. Unencrypted Web access is not an option.

Security precautions are available on the Linux server to ensure user access is restricted to only information that is required. Standard Linux operating system capabilities allow creation of login/password sets and permission groups allowing only read and/or write access to the partitions of the server that are needed by each user.

Avaya Communication Manager allows the administrator to modify the permissions associated with a login. The system maintains default permissions for each level of login, but the customer may want to further restrict a login, or make sure the defaults are appropriate for the user. The default values for these fields vary based on the login type.

As an option Avaya recommends deployment of technology from one of our “Premier” level Avaya Developer Connection (DevConnect) program partners. The ETM® System from SecureLogix Corporation could be deployed across all inbound PSTN circuits to detect call type and enforce security policies

preventing outbound modem calls by employees to unauthorized Internet Service Provider (ISP) accounts. ETM® also detects call type and enforces security policies preventing unauthorized inbound modem calls to modem-enabled workstations and other devices.




Avaya is not aware of any other solution providing comparable telecom network security capabilities.

The ETM System monitors all outbound calls, and screens for voice network security violations. The system’s Voice Firewall application contains patented technology to distinguish call type (voice, fax, modem, etc.) in real-time. The firewall can enforce telecom network admission control rules based on a multitude of call characteristics, such as source, destination, time, duration, and call type. A global firewall rule can block and alert all outbound modem calls originating from within the corporate electronic security perimeter. If a number of authorized dial-up connections are to be allowed, voice firewall rules can log, alert, and prevent “outbound modem” type calls to a black list of local and national ISP destinations. Alternatively, a firewall policy could be used to simply block all “outbound modem” type calls, except for authorized modem calls to a white list of authorized callers, or paired source/destination numbers. Any of these approaches will alert and prevent all unauthorized Internet session calls to private ISP accounts.

The ETM System monitors all inbound calls, and screens for voice network security violations. As mentioned above, the system’s Voice Firewall application contains patented technology to distinguish call type (voice, fax, modem, etc.) in real-time. The firewall can enforce telecom network admission control rules based on a multitude of call characteristics, such as source, destination, time, duration, and call type. A global firewall rule can block and alert all inbound modem calls to workstations and devices inside the corporate electronic security perimeter. If authorized dial-up connections to certain devices are to be allowed, voice firewall rules can allow and log calls to a white list of paired source/destination numbers, while alerting and preventing all other “inbound modem” type calls. Further, a SecureLogix AAA Services appliance could be deployed to enable 2-factor authentication for authorized calls into modem-enabled systems for remote access and maintenance. Either approach will alert and prevent all unlawful data network access by outsiders penetrating through modem-enabled corporate phone lines connected to LAN/WAN accessible workstations and other equipment.

1.6.3 Disruption of Services VoiceCon requires that the proposed IPTS solution be secure against disruption of services. A minimum, the vendor will should:

• Provide built-in DoS resiliency for all components processing signaling and audio.

• Provide embedded or compatible third party firewalls, IDS/IPS systems, or anti-DoS systems will be available.

• Support DoS detection and mitigation capabilities in network switches

• Provide a solution for malformed or “fuzzed” packets

• Provide protection for key supporting infrastructure services, such as TFTP, DHCP, DNS, etc. will be provided.


Briefly describe any embedded features/functions in the proposed IPTS solution that will reduce probability of telephony services disruption due to Denial-of-Service (DoS) attacks and address each of the above listed items in your response.


and each station-to-station call.



Avaya Response:

VoiceCon can isolate the S8X00 servers on their own network that can be physically independent from VoiceCon’s general network or even logically isolated in its own VLAN using a non-routable "Private Network", where nobody needs to know their IP address. This provides additional protection for servers and minimizes the risk of Denial of Service (DoS) and other threats. Avaya IP Telephones can also be placed on a separate VLANs then the PC plugged into the IP phone using standards based protocol such as 802.1ab (LLDP), LLDP-MED, or DHCP for VLAN assignment.

The proposed Avaya Communication Manager environment is open and standards based – essentially network infrastructure agnostic, for real-world flexibility. The protection of supporting infrastructure services such as HTTP (we have moved away from TFTP), DHCP, DNS, etc is best handled by the network infrastructure elements. In addition to traditional features like access control lists and exception packet auditing, many network infrastructure vendors now offer specialized features integrated into their network infrastructure products to provide additional layers of defense around these IP Telephony supporting services.

1.6.4 Theft of Services Vendor Response Requirements

Briefly describe any embedded features/functions in the proposed IPTS solution that will identify the incidence of toll fraud and other types of Long Distance toll service abuse/misuse (e.g. LD voice calls on fax lines) in real-time, and alert and/or block such activity to reduce financial losses.

Avaya Response:

Avaya’s Call Detail Recording (CDR) feature will record information on incoming, outgoing, and tandem calls for each trunk group that you administer for CDR,

including auxiliary trunks. The system records information on each trunk-group call

You can also request that CDR record information on:

¾ Temporary signaling connections (TSCs) that involve trunks

¾ Calls that use loudspeaker paging

¾ Calls to which account code dialing or a feature access code (FAC) apply

¾ Ineffective call attempts

If you request that the system record information in ineffective call attempts, you greatly increase the number of calls that the system records. However, the request to record ineffective call attempts can also help you to increase security, because the system records call attempts that are blocked because of insufficient calling privileges. Information on ineffective call attempts can also show you that your users cannot make


calls because all the trunks on your system are busy.

¾ The audio service link calls that the switch uses for IP Softphone that are set up as telecommuter IP Softphone.



An IP Softphone can use one audio service link to make many short calls. The system shows these many short calls as one long call on the CDR reports.

Avaya protection of customer networks does not end with maintenance services. Toll fraud prevention and security of your vital communication networks are areas of concern that you need to know are covered by an experienced and proficient provider.

When you detect telecommunications fraud is being perpetrated, you can contact the PBX Security Services Team who will secure the avenue/path of fraud. Locking down the avenue/path where active fraud is being committed will assure fraud is mitigated and prevent future occurrences. Incident Response is available 24 hours a day, 365 days a year.

1.6.5 Restricted Calls Vendor Response Requirements

Briefly describe any embedded features/functions in the proposed IPTS that will identify telephony/fax spam, harassing calls, and other types of restricted calls (e.g. bomb threats, threatening calls, calls to/from restricted numbers) in real-time, and alert and/or block such activity to reduce damages and legal exposure.

Avaya Response:

Communications Manager has Malicious Call Trace (MCT) as a standard feature to track malicious calls. Both users and attendants can track malicious calls, and display information that identifies the source of the call. The user or attendant can share the information with emergency personnel on a tandem server. The personnel on the tandem server can continue to trace the call. MCT also supports a voice recorder that records the malicious call.

Blocking calls can be handled via numerous features standard on Communication Manager, some are described below.

Called party restrictions

Called party restrictions define the privileges for telephones that receive inbound calls. If you do not need to restrict telephones that receive inbound calls, assign a COR with the Called Party Restriction field set to none.

Calling party restrictions

Calling party restrictions define the privileges for telephones that make outbound calls. If you do not need to restrict telephones that make outbound calls, assign a COR with the Calling Party Restriction field set to none.

You can use called/calling party restrictions for:

¾ Unrestricted telephones

¾ Trunk groups

¾ Terminating Extension Groups (TEGs)

¾ Uniform Call Distribution (UCD) groups

¾ Direct Department Calling (DDC) groups

¾ Data modules

¾ Attendant groups

¾ Individual attendant extensions




All-Toll restrictions and TAC-Toll restrictions

Toll restrictions prevent users from placing public-network calls to certain toll-call numbers. You assign toll restrictions to outgoing trunk groups on the Trunk Group screen. You disable Trunk

Access Code (TAC)-toll restrictions for specific outgoing trunk groups on the Trunk Group screen.

Origination restrictions

You can use origination restrictions to prohibit users from originating calls. These users can still receive calls.

Outward restrictions

You can use outward restrictions to prevent users from placing calls to the public network.

These users can still place calls to other telephone users, to the attendant, and over tietrunks. If necessary, an attendant or an unrestricted telephone user can extend a call to anoutside number for an outward-restricted telephone user.

Calls that come into a trunk are denied if the:

¾ Calling Party Restriction field on the Class of Restriction screen is set to outward

¾ Calls use the Automatic Alternate Routing (AAR) or the Automatic Route Selection (ARS) feature

Inward restrictions

You can use inward restrictions to allow users to receive only internal calls. With inward restrictions, users at assigned telephones cannot receive:

¾ Public network calls

¾ Attendant originated calls

¾ Attendant extended calls

The system checks only the COR of the originally called telephone, unless you administer a three-way COR on conference calls and transfer calls. The system routes denied calls to:

¾ Intercept tone

¾ A recorded announcement

¾ The attendant, for Direct Inward Dialing (DID) calls

Fully restricted service

With fully restricted service, users cannot make or receive public network calls. Users who have fully restricted service cannot use authorization codes to deactivate this feature.

COR-to-COR restrictions

You can restrict calls from one COR to another COR. You can use COR-to-COR calling restrictions to prohibit user access to specific telephones or specific trunk groups, such as CO trunk groups. Any or all trunk groups can be in a trunk-restriction COR. The system routes restricted calls to intercept tone.




1.6.6 Confidentiality and Privacy (Packet Sniffing) VoiceCon requires that the proposed IPTS solution provide for a high degree of confidentiality and privacy, including:

• Support for standards such as IPSec, TLS, and SRTP.

• Encryption for all public (to the LAN) traffic must be supported. This includes traffic exchanged between the call processor and media gateway.


Briefly described any embedded features/functions in the proposed IPTS that will preserve communications confidentiality and privacy, including the standards listed above. Indicate if control signaling and/or bearer communications signaling is encrypted at the call control, voice client, and media gateway elements to counter packet sniffing attempts:

Avaya Response:

Comply. The embedded application encryption features of the proposed solution provide encryption of the call control (signaling), voice client both H.323 and SIP (signaling/media), and media gateway (signaling/media) elements to protect user privacy. Avaya has openly embraced the National Institute of Standards and Technology (NIST) FIPS 197 Advanced Encryption Algorithm (AES). The Avaya implementation of the NIST FIPS 197 Advanced Encryption Algorithm (AES) uses a 128bit encryption key. Avaya utilizes the ITU H.235.5 (for H.323 end-points) and TLS (for SIP end-points) to secure information in the signaling between IP end-points and Avaya Communication Manager/SIP Enablement Services (SES). TLS is utilized to secure the connections between Avaya Communication Manager, SES, and the media gateways. As it regards media encryption, the selection of an underlying encryption algorithm has to stand up to the highly demanding requirements of real time media streams which include scalability, end-point processing capacity and high tolerance to packet loss and re-ordering. The encryption-processing overhead must minimize CPU cycles and result in minimal latency and jitter. In non-technical terms this means that the differences in voice quality resulting from the encryption/decryption process will be imperceptible to the end-user. Testing has shown a minimal 1-2% increase in end-point processor utilization and less than 5ms added end-to-end latency. Avaya supports two frameworks for media encryption, Avaya Media Encryption and the IETF SRTP standard (RFC 3711).

Avaya Media Encryption adds no additional IP overhead per packet. SRTP may add additional overhead depending on the standards defined Message Authentication and Integrity option chosen (HMAC80 adds 10 additional bytes per packet, HMAC32 adds 4 additional bytes per packet) and offers the potential for standards based interoperability with third party vendors as we have demonstrated at past industry trade shows.




Avaya’s implementation of Media Encryption offers the following benefits:

¾ The ability to support end-to-end media encryption throughout the Avaya IP telephony endpoint portfolio.

¾ The flexibility to determine the framework that is used. The Avaya Labs developed Avaya Media Encryption using AES encryption or the IETF Secure Realtime Transport Protocol (SRTP) standard using AES encryption.

¾ The ability to configure the usage preference of the framework and its supporting encryption/authentication methods.

Avaya is committed to media privacy, and has taken the lead in offering media privacy. Avaya Media encryption capabilities have existed for well over six years, since DEFINITY Call Processing release 10. Media Encryption has been operating in real world mission critical environments and is field proven.

1.6.7 Physical Interfaces Vendor Response Requirements

Are there separate physical network interfaces to IPTS administration, control, and voice transmission signaling functions?

Avaya Response:

Comply. To provide the most secure environment that is possible for the system, network access should be divided into separate zones of control.

¾ One VLAN can be administered for administrative traffic, one for call signaling, another for voice bearer traffic, and so on.

¾ Layer 3 boundary devices (routers, layer 3 switches, and firewalls) can be administered to enforce the corporate security policy on traffic that is destined for the Avaya S8X00 Server, Avaya Media Gateways, or adjuncts.

Packet filters can permit administrative access only from an administrator’s PC and to deny access from the Avaya S8X00 Server or its gateways to the corporate LAN while allowing call signaling and bearer traffic from all IP Telephones appropriate access.

In addition, the Separation of Bearer and Signaling (SBS) feature of Avaya Communication Manager provides a low cost virtual private network with high voice quality for VoiceCon, and supports cost avoidance by eliminating some of the requirements for private leased lines.

The SBS feature supports:

¾ QSIG private networking signaling over a low cost IP network.

¾ Voice (bearer) calls over the public switched network.

¾ Association between QSIG feature signaling information and each voice call.




Recommendations for Network Security

Avaya recommends that these servers be located behind a firewall. Where this firewall is located with respect to other LAN components must be designed on a case-by-case basis. Avaya Professional Services can assist VoiceCon in configuring their network for both security and optimal IP Telephony operation.

Other vendors also specialize in this type of consulting. Owners are advised to seek assistance if internal staff is not trained in these areas. Security oversights that arise from negligence, ignorance, or the pressures of schedule or budget are all equally usable by hackers. Malicious activity is a moving target, and what is safe today might not be safe tomorrow. Avaya is committed to providing appropriate secure solutions

for its products, and to continuously monitoring evolving security threats. Avaya S8X00 servers are appropriately secure against the known threats and hardened to be resilient against threats to come. Avaya responds quickly should new threats appear and proactively notifying customers via security bulletins which classify the vulnerability and offer expected remediation time frames.

Consult these resources for the latest security information:

¾ Your Avaya account team

¾ The Avaya Security support Web site: http://support.avaya.com/security

1.6.8 Root Access Vendor Response Requirements

Is there direct Root access to the IPTS common control, and does the proposed IPTS solution conforms to the following design attributes:

• Disablement of non-secure management interfaces such as telnet bydefault.

• No installation of any default administrative or root passwords.

• Logging of all activity for administrative or root access.

Avaya Response:

On a Linux system the highest level of administrative access isknown as "root". Direct login to a root level account is notpermitted on an Avaya S8X00 Server. Administrative commands that require root level permissions are handled via a "proxy" program that grants specific access to specific accounts for specific programs with extensive logging of these delegated commands. The ability to obtain full root level access is granted only in very special circumstances. This policy helps protect the integrity of core software. All insecure management protocols are disabled by default on management interfaces, examples include Telnet replaced with SSH, HTTP replaced with HTTPS, SNMPv3 (if enabled) replacing previous versions of SNMP. There are no "default" administrative passwords. Avaya accounts used for system maintenance utilize a one-time password challenge/response authentication process which is unique for each customer installation. Avaya Communication Manager 4.0 and greater has added extensive auditing (logging) capabilities throughout the Avaya Communication Manager application, HTTPS and CLI interfaces. A centralized external syslog server is now customer configurable which allows for easier aggregation, archiving, and manipulate of logged information in customer environments.


of



Avaya Communication Manager 4.0 and greater have also added the capability of defining unique administrative profiles (roles) for both the Avaya Communication Manager application and HTTPs interface, so that administrators are given only those authorization privileges required to carry out their core job function.

1.6.9 Miscellaneous Security Requirements Vendor Response Requirements

VoiceCon requires that the proposed IPTS solution provide the following general security features:

• A patch management process and system must be available.

• A secure alternative to TFTP (whose files can easily be sniffed) must be provided.

• Support of TCP and authentication should be provided if SIP is supported.

• Firmware loads for IPTS phones will be signed to insure authenticity.


Confirm that the proposed IPTS solution satisfies each of the listed general security features:

• Patch management

• TFTP alternative

• Signed firmware loads

Avaya Response:

Comply. Avaya has a well defined patch management process. The http://support.avaya.com/security website contains two whitepapers which clearly detail the patch management process for our customers. These whitepapers are titled, "Avaya’s Security Vulnerability Classification" and "Avaya’s Product Security Vulnerability Response Policy". The whitepapers provide an explanation of our security vulnerability classification process (High, Medium, Low) and detail how customers can sign up to be proactively notified of security vulnerabilities or status changes to existing bulletins. Vulnerability classification determines the documented remediation timeframe for the Avaya product impacted by the vulnerability. In the period before remediation is available, workarounds are detailed in the bulletins to help customers mitigate the threat. Beginning with Avaya Communication Manager 4.0, Avaya has introduced a separate patching cycle for security vulnerabilities termed security service packs. This separation of Communication Manager service packs from security (OS) service packs provides our customers greater flexibility in their patch management process.

Avaya has moved away from insecure management and file transfer protocols which expose authentication credentials and on-going session communications which expose confidential information. These insecure protocols have been disabled by default and replaced with their standards based secure alternatives. In particular, TFTP and FTP both clear text protocols, have been replaced with Secure Copy (SCP) and Secure FTP (SFTP).

The Avaya SIP Enablement Services (SES) server exceeds the requirement of standard SIP/TCP by the establishment of a secure SIP/TLS session between SIP components. Avaya follows the standards based methods authentication across this TLS link detailed in RFC 3261 - SIP: Session Initiation Protocol termed digest authentication for both registration and when appropriate, proxy authentication.




Avaya IP Telephone firmware is digitally signed and verified by the IP Telephone after download to prevent the introduction of tampered firmware into the device. The digital signature for the software code image is generated using the Digital Signature Algorithm (DSA) of the FIPS Digital Signature Standard, which uses the FIPS Secure Hash Standard.

Software Release Management (optional)

As an option Avaya offers Software Release Management (Patch Management for IP Telephony)

Avaya Software Release Management provides a solution for the identification and implementation of product update notices. Product update notices include product correction notices (PCNs), software updates, and firmware updates. The solution provides Avaya expertise and automation to implement recommended updates that maximize performance of applications and minimize the risk of potential service disruptions. Avaya manages the identification, risk assessment, schedule and implementation for the updates.

The initial release of Software Release Management provides support for Avaya's IP Telephony application which includes Communication Manager software, the Linux-based Server platforms and Media Gateways.

Customers will need to provide a managed, switched, Fast Ethernet LAN connection to each Avaya S8XXX Server and Media Gateway to support remote delivery of this offer.

Avaya will offer the Software Release Management solution as an affordable alternative for customers who want Avaya as their single point of accountability to manage the implementation of product updates. The Software Release Management solution allows customers to maintain authority while leveraging Avaya’s expertise to manage change.

Avaya uses our new Enterprise Service Platform to enable remote delivery of this solution. Our capability allows us to quickly map each released update to identify customer sites and systems that require a configuration update. Our skilled delivery team documents the benefits and coordinates the authorized updates using Avaya best practices and specialized tools. The automated, proactive process easily scales to support multiple customer sites and products to ensure they are operating at the optimum software and firmware release.

Software Release Management is for customers who want Avaya to manage the risk assessment, schedule and update process for the supported products. These customers have limited internal experts or tools to help them assess when to implement an update and do not want to invest in resources or tools to support this process. These customers will benefit from Avaya’s ability to create inventory records, recommend minimum release versions, notify them of new product updates including impact assessments and coordinate the update schedule. Customer participation will still be needed to review Avaya’s recommendation for the update and authorize the final decision to implement the change. The solution includes the following service elements:

¾ Implementation Coordinator

¾ Inventory Management

¾ Product Update Notification

¾ Product Update Schedule

¾ Product Update Qualification

¾ Product Updates

¾ Monthly Reports

¾ Notification Objective

¾ Customer Web Portal



Part 1 - Section 2 - IPTS Network Port Capacity

2.0 IPTS Network Port Capacity Requirements The proposed IPTS must be capable of supporting port capacity requirements for the HQ facility and remote branches. It must also be capable of supporting future VoiceCon growth requirements at HQ and RO facilities.

Avaya Response:

Comply. The recommended system meets and/or exceeds all requirements.

2.1.0 Port Capacity Requirements The equipped port capacity of the proposed VoiceCon HQ IP Telephony System at time of installation and cutover must support of a mix of IP telephones, analog telephones, facsimile terminals, modems, central office trunk circuits (analog and digital) for local and long distance services, and private network IP trunk circuits).

Avaya Response:


In support of general communications requirements, VoiceCon facilities will have a sufficient number of wiring closets distributed throughout each facility to satisfy ANSI/EAI/TIA 569 structured cabling specifications for voice and data communications. Wiring closets will be interconnected based on requirements of the selected system. The entrance facility (trunk connect panel), main telecom equipment room, and Main Distribution Frame (MDF) for each facility are located off the entrance lobby. It will be the responsibility of the contractor to provide all cross connects between labeled 110 terminal blocks in each wiring closet and the demarc or "smart jack" and their equipment. The following sections describe the port capacity requirements for each of the VoiceCon network locations. Satisfying these stated port capacity requirements is a MANDATORY requirement

Avaya Response:

Comply with explanation; Avaya makes one cross connect at the Main IDF (PBX Room) in the price of the proposal. If additional cross connects are required Avaya will be able to quote this after a site visit.

Keep in mind however that any and all Cut Sheets containing House Cable Pair assignments from the MDF to IDF as well as Cable pair assignments from the House Cable pair to the Station wire number in all of the IDF's would also need to be provided by VoiceCon or its wiring vendor.

As for any Data Patch Panel activities for VoIP in Data closets, since we generally plug our Terminal into an active Data jack and than plug the PC into our VoIP set whom ever the Vendor is that is installing the Data Equipment would also be installing all Data Patch Panel Patch Cords at all IDF locations.

2.1.1 CHQ Facility The CHQ campus includes two buildings, each a four-floor facility, with station equipment uniformly distributed within and across the four floors of the building. There are five (5) wiring closets per floor and one (1) main equipment room on the first floor of each building.

Avaya Response:

Read and understood




2.1.2 RO Facility The RO facility has two floors with station equipment uniformly distributed within and across each floor of the building. There are five (5) wiring closets per floor and one (1) main equipment room on the first floor.

Avaya Response:

Read and understood

2.1.3 SO Facility The SO facility has one floor with uniformly distributed station equipment across building. There are two (2) wiring closets and one (1) main equipment room.

Avaya Response:

Read and understood

2.2 Port Requirements VoiceCon requires that the proposed IPTS communications solution be designed to support the following equipped port capacity requirements in the following tables with wired capacity for 50% growth. The definition of “equipped capacity” is all necessary system hardware and software components installed and working at time of initial system installation based on the stated requirements of this RFP. The definition of “wired for capacity” is the capability for system expansion (call processing, switched connections, and port capacity) with the addition of port interface cards or media gateway boards, only.

VoiceCon will engineer its WAN trunk circuits to support compressed voice traffic (G.729A algorithm voice codecs) across its facilities. In addition to the following trunk circuit requirements any additional trunk services necessary to support the proposed IPTS, e.g., local survivability requirements, must be identified, configured, and included in the pricing proposal. Necessary common equipment must be included in the system configuration and pricing proposals and identified as such.

Avaya Response:

Comply. The recommended system meets and/or exceeds all requirements. The S8500C supports up to 2400 IP endpoints with 800 trunks.

2.2.1 CHQ Station/Trunk Port Requirements 2.2.1.1 CHQ Building 1

Station Equipment • Analog devices (VoiceCon provided): 75

o 2500-type telephone instruments: 55

o Modems: 15

o Facsimile terminals: 5

• IP terminals (See Section 4) 675

o Economy desktop instrument 25

o Administrative desktop instrument 50

o Professional desktop instrument 425

o Executive desktop instrument 75

o ACD agent/supervisor telephone instrument 75




o Attendant soft consoles 3

o Audio conferencing units 22

Trunk Circuits • Local Service

o GS/LS circuits 48

o T1-carrier (PRI) circuits 3

• Long Distance


• Inbound ACD


Avaya Response:


2.2.1.2 CHQ Building 2

Station Equipment • Analog devices (VoiceCon provided): 50


o Modems: 10







o ACD agent/supervisor telephone instrument 50



Trunk Circuits

• Local Service

o GS/LS circuits 32


• Long Distance


• Inbound ACD





Avaya Response:


2.2.2 RO Station/Trunk Port Requirements Station Equipment

• Analog devices (VoiceCon provided): 25


o Modems: 5









Trunk Circuits

• Local Service

o GS/LS circuits 8


• Long Distance


Avaya Response:


2.2.3 SO Station/Trunk Port Requirements Station Equipment

• Analog devices (VoiceCon provided): 5


o Modems: 3












Trunk Circuits

• Local Service

o GS/LS circuits 8


• Long Distance


Avaya Response:




Part 1 - Section 3 - Port Interface and Traffic Handling Capacity Requirements

3.0 Port Interface and Traffic Handling Requirements The proposed IPTS must support a variety of peripheral ports and switched connections. In addition to IP endpoints it is required to support traditional circuit switched analog stations and analog & digital trunk circuit interfaces. The common equipment (port interface carriers, media gateways) must be supported in a distributed topology using VoiceCon’s LAN/WAN for transmission and switching of communications and control signaling. Common equipment must be located at CHQ, RO, and SO facilities.

Any and all port interface cabinets/carriers designed to support traditional analog and digital interface ports should include an integrated TDM bus backplane traffic engineered to support non-blocking switch network access for all peripheral endpoint connections. Transmission and connections between all TDM buses must also be traffic engineered to operate in non-blocking mode. A center stage switch network, if equipped, must also be traffic engineered for non-blocking access.

Media gateway equipment should be designed and configured to support a 4:1 ratio between IP peripheral endpoints (line station and trunk circuit) and media gateway channels used to connect to non-IP ports.

For common equipment configuration design purposes assume the following voice communications traffic volumes:

User line station (analog and IP) Attendant console position ACD/Supervisor call center agents Voice mail port traffic All trunk circuit traffic

12 CCS at busy hour 36 CCS at busy hour 36 CCS at busy hour 36 CCS at busy hour 36 CCS at busy hour

The assumed system voice communications traffic mix should be 40% station to station calls, 30% incoming trunk calls, 30% outgoing trunk calls.


The proposed system IPTS be designed and engineered to support the above traffic assumptions. Confirm you have satisfied this requirement.

Avaya Response:

Comply.

3.1 Common Equipment The proposed IPTS solution must support a variety of peripheral ports and switched connections. Although it is not required to support traditional digital voice terminal equipment, the IPTS must support analog communications devices and PSTN trunk circuits. Switched connections involving non-IP ports may be handled using a circuit switched network, media gateways/Ethernet switches, or a combination of both.


Briefly identify by make/model the proposed common equipment and describe each type of equipment housing used to support ports, circuit card interfaces, media gateway boards, and other required equipment. Specifically discuss in the response housing size (H x L x W), weight (standard common assembly), fan cooling units, power supply requirements, number of usable port card slots per carrier/chassis, and all embedded hardware components.

Avaya Response:

Comply. The proposed solution will support analog, IP, SIP as designed. Traditional digital voice terminal equipment can be added via a circuit pack addition.




The Avaya S8500C Server can support up to a system maximum of 15,424 simultaneous circuit switched calls or 242 per Gateway. There are 238 time slots for voice and data per G450. For IP to IP calls this limitation does not apply.

DHCP (Dynamic Host Configuration Protocol) server functionality enables you to automatically assign IP addresses and other network parameters to remote stations not configured with static network parameters. A pool of allocated addresses and parameters is created on the server.

The remote station, on network login, requests network parameters from the DHCP server. The DHCP server provides the remote station with parameters such as IP address, subnet mask, default gateway, and Domain Name Server (DNS) information.

The Avaya G250/G350/G450 Device can act as a DHCP server both for devices physically connected to the Avaya G250/G350/G450 Device and for other devices on the same network.

The G450 has eight expansion slots for MM-type media modules.

Please see 1.3.1.1.1 for additional environmental information

3.1.1 Universal Card Slots VoiceCon prefers that the proposed common equipment be based on a universal card slot design for all TDM port interface circuit cards.


Confirm that your proposed system satisfies this requirement.

Avaya Response:

Comply all of the G450 & G350 Avaya Media Gateways support the same Media Modules.

3.1.2 Common Equipment Redundancy VoiceCon requires an IPTS that satisfies a very high degree of reliability and services availability. To achieve this goal IPTS common equipment should include a significant number of redundant design elements to minimize the affects of single points of failure.


Confirm if the proposed common equipment includes any or all of the following redundant common equipment elements, and include a brief description of the level or type of redundancy provided for each element.

• Service circuits

• I/O interfaces

• Media gateway boards

• Control signaling interfaces

• TDM bus backplane

• Inter-TDM bus switch network connections

• Center stage switch network (if applicable)

• Internal power supply




Avaya Response:

One of the biggest concerns you may have when deploying IP telephony is with the up-time of the solution. Up-time depends on two things - reliability of the components and the resiliency of the design. Reliability of the Avaya components is a given with the tried and true software and hardware backed by hundreds of thousands of hours of MTBF (Mean Time between Failure), but in the IP telephony world where this business critical application has to travel through a multi-vendor network to be effective, resiliency becomes a key differentiating feature.

Avaya offers a multi-layered approach towards resiliency as illustrated in the graphic below and explained in the following paragraphs.

First Level (Optional in VoiceCon’s design)

Avaya offers stateful server duplication, using the Avaya patented Memory Shadowing, where only one S87XX-Series Server is running the entire system of up to 36,000 stations and 8,000 trunks. The non-active redundant server is synchronized real-time (either via a dedicated fiber or across an IP network) to the active server, providing virtually imperceptible fail-over in the event of major component failure in the active server. Not only are the calls maintained during failover but call control (for example, the ability to transfer a call) is preserved as well. In addition, any calls that may be queued in a call center or even calls that are being set up are maintained, including the continuation of processing digits being entered.





Second Level (Optional)

Avaya offers multiple/dual-homed interfaces from the servers to the media gateways (or Port Networks), where each server and each media gateway can be connected to duplicate Ethernet switches at the same time. This provides redundancy for cases of component failure (server, Ethernet port, IPSI or IP Server Interface) and network segment failure.

Third Level (Included in VoiceCon’s design)

Avaya provides separate signaling and bearer (voice) interfaces in the form of C-LAN (Control LAN/H.323 Gatekeeper functions) modules for signaling and DSP resources for bearer traffic. This separation provides resiliency in the form of protection from a single interface failing or being attacked maliciously. This separated signaling/bearer model is a significant differentiator in the Avaya architecture. End-points and remote Media Gateways register with one of the multiple LSPs to receive telephony services. Because the endpoints and computers talk to one of those multiple signaling channels instead of talking directly to the servers, VoiceCon can isolate the S8500-Series Servers in their own network that can be physically independent from VoiceCon's network or even logically isolated in its own VLAN using a non-routable "Secret Network", where nobody needs to know their IP address. This provides unmatched protection for servers, mitigating the need for constant virus updates and minimizing the risk of Denial of Service (DoS) and hacker attacks. Architecturally, redundant C-LAN cards allow connections to different blades on the core Ethernet switches, or to completely different Ethernet switches, improving the resiliency towards network segment failure and DoS attacks. If attacked, the Gateway is equipped with sophisticated discard strategies and recovers shortly after the attack is completed. If a Gateway does become unavailable, the endpoints are pre-programmed with a list of alternate gatekeepers, so they automatically look for an alternate Gateway for service. Load sharing is done automatically between all the Gateways in a network region. Separating the signaling channels from the bearer channels also allows VoiceCon to take advantage of using the IP network for signaling traffic and using the PSTN for bearer (voice) traffic should the IP network be incapable of supporting the offered load. This is yet another significant differentiator between Avaya and some of our competitors.

Fourth Level (Optional recommended for large Enterprises over 2400 endpoints)

Avaya offers the Enterprise Survivable Server(s) (ESS) solution which is used for catastrophic failures in VoiceCon’s enterprise. The ESS solution allows up to 63 ESS (backup) servers at key locations throughout the enterprise which can be activated hierarchically. An ESS server is capable of taking over for the entire enterprise or, if needed, just a portion of the enterprise in the event that some type of outage has occurred. This is an automatic failover that is pre-programmed and occurs without any additional configuration changes or administration on the ESS servers. All configuration modifications are made to the primary active server and Avaya Communication Manager processes the duplication of these changes automatically to all other Survivable Servers in the network to ensure consistent user features for all sites covered by a survivable server.

Fifth Level (Included in VoiceCon’s design)

Avaya provides redundancy for Announcements and Queue Music, which not only increases the availability of these imperative functions, but is also designed so that in most instances these audio sources are played locally closest to where the call came into the network, making sure that valuable WAN resources are not consumed by streaming this audio over the WAN.



Sixth Level (Optional)

If VoiceCon has diverse WAN connectivity to remote sites, you can take advantage of the proactive monitoring and redirection capabilities of Avaya Converged Network Analyzer (CNA) with Adaptive Path Control (APC) to get sub-second failover when WAN failures or brownouts occur. CNA monitors application performance, WAN connectivity (packet loss, jitter, and delay), bandwidth utilization, etc., and can "rescue" specific applications (not only limited to VoIP), certain hosts, or entire sites as needed. This is a night and day difference from traditional routing protocols which can take a minute or more to recover and can only reroute based on an actual failure with no mechanism to deal with the equally prevalent “brownout” defined as a degradation in service or performance below a specific application’s thresholds. CNA works with routers from Juniper, Cisco and many other vendors.

Seventh Level (LSP Included in VoiceCon’s design)

Avaya provides local processing capabilities to protect against WAN outages. Local processing can be offered in two variations – full feature for all models of gateways (Enterprise Survivable Server and Local Survivable Processor) and basic survivability (Standard Local Survivability) for the small location gateways. ESS/LSP survivability can be provisioned for up to 2,000 users and offers the following benefits

¾ Zero administration of remote site gateways, since all configurations are administered once at the central site and flow out to the remote locations. This minimizes support costs and eliminates problems associated with having to administer changes in multiple places.

¾ The telephony functionality does not rely on a specific brand or network features of the router, which eliminates a single point of failure, and increases the reliability of the telephony services.

¾ No need to worry about upgrading a voice gateway to support a patch that addresses a data problem, with all the possible implications to voice down time.

¾ Ability to provide all the features that are available in normal operation which includes call center agents, locally sourced announcements, IP softphone support, E-911 and all end user phone features.

¾ Reduced long term cost of ownership, since historically, customers tend to upgrade their router hardware much more often than they upgrade voice systems. With some IP telephony systems from some suppliers, every time VoiceCon upgrades your routers you would be required to re-purchase the "extra voice blade in the router" making it a recurring capital expenditure, rather than a one time cost.

The S8500 is a single processor our redundancy comes with the Local Survivable Processor design presented in other portions or this RFP. An optional external power supply for the G450 Media Gateway is available.


(DHCP)



3.2.1 IP Station Discovery How do IP communications devices learn about their voice VLAN, including IP addresses, default gateways, call controller, TFTP server, QoS settings, VLANs, and other parameters. Does the proposed system solution employ proprietary protocols for IP communications devices to learn their voice VLAN or is an industry standard, such as Dynamic Host Control Protocol (DHCP) used?

Avaya Response:

Avaya IP Telephones can use Dynamic Host Configuration Protocolstandard to learn their IP addresses, default gateways, call controller, TFTP server, QoS settings, VLANs, and other parameters. Avaya can use a special option, Option 176, to pass these values. Avaya has done significant testing of and had good success with Option 176 on the Microsoft Windows 2000 DHCP server and the ISC DHCP server (common on Linux and UNIX platforms).

Discovery and Registration Process to the Gatekeeper

3.2.2 IP Station Power over Ethernet (PoE) VoiceCon requires that the power option to support IP telephones conform to IEEE 802.3af Power over Ethernet (PoE) standards.


Confirm that the proposed IPTS solution supports the IEEE 802.3af specification for in-line of IP telephone equipment. Describe current, future and retrospective compatibility of all proposed equipment. If 802.3af is not supported, identify the PoE implementation being proposed.

Avaya Response:

Avaya, as an industry leader and visionary, has actively participated in the development of the IEEE 802.3af standard since its inception several years ago. In June of 2003 the IEEE approved 802.3af now known as "802.3af-2003."




All IP telephones currently being sold by Avaya are compatible with the standard. We remain committed to designing our products to be compliant with the specifications of the standard as is demonstrated by our leadership position within the committee responsible for the recommendation.

Our strength lies in the number of different options that we can provide compared to our competitors. These include:

¾ Avaya 4600 and 9600 Series IP Telephones that are designed to comply with the specifications of the IEEE 802.3af standard

¾ Avaya Mid-Span Power Solutions (a patch panel type device that adds 802.3af power capability to non-802.3af compatible Ethernet switches - available in 1, 6, 12 and 24 ports)

¾ Avaya P333T-PWR layer 2 switch designed to support 802.3af power

¾ Power over Ethernet (PoE) blade for the C460 multi-layer chassis based switch designed to support 802.3af power

¾ The 1151B1 individual external power supply and the 1151B2 which has built in battery backup for IP phones

¾ An adapter which allows customers to power newer Avaya telephones from certain Cisco switches that only provide proprietary power

¾ The Avaya AP3 (2nd generation), AP4, AP5, and AP6 Access Points that can be powered by an IEEE 802.3af compatible device

Avaya IP telephones have been and continue to be compliant since May of 2002. Avaya is committed to developing standards based solutions that allow the customer the flexibility to choose solutions based on their business needs as opposed to forcing them to one manufacturer.

3.2.3 IP Station QoS Vendor Response Requirement:

Describe the proposed IPTS solution’s capabilities to provide Layer 2 and Layer 3 QoS to IP stations to ensuring end-to-end quality of service. Include in the response what industry standards are deployed.

Avaya Response:

QoS for voice packets is obtained only after a Class of Service (CoS) mechanism tags voice packets as having priority over data packets. IEEE 802.1p/Q at the Ethernet layer (Layer 2) and DiffServ Code Points (DSCP) at the IP layer (Layer 3) are two standards based CoS mechanisms that are used by Avaya products. These mechanisms are supported by the IP Telephone, the S8300 Server. Although TCP/UDP source and destination ports are not CoS mechanisms, they can be used to identify specific traffic, and can be used much like CoS tags. Other non-CoS methods to identify specific traffic are to key in on source and destination IP addresses and specific protocols, such as RTP.

Class of Service refers to mechanisms that tags traffic in such a way that the traffic can be differentiated and segregated into various classes. Quality of Service refers to what the network does to the tagged traffic to give higher priority to specific classes.




If an endpoint tags its traffic with Layer 2 802.1p priority 6 and Layer 3 Differentiated Services Code Point (DSCP) 46, for example, the Ethernet switch must be configured to give priority to value 6, and the router must be configured to give priority to DSCP 46. The fact that certain traffic is tagged with the intent to give it higher priority does not necessarily mean it will receive higher priority. CoS tagging is not effective without the supporting QoS mechanisms in the network devices.

3.3 Multi-Party Conference Calls The proposed system must be able to support six party add-on conference calls among IPTS stations and off-network stations. The system must also support a minimum of three (3) off-network stations per multiparty conference call when required. The HQ IPTS must support a minimum of 20 simultaneous multiparty add-on conference calls (up to six parties per conference) and the RO IPTS a minimum of 10 simultaneous multi-party add-on conference calls (up to six parties per conference)


Briefly explain how multi-party add-on conference calls are handled if:

1) All parties are on-network IP stations;

2) There is a mix of on-network IP and off-network stations.

The explanation should identify any and all hardware and software requirements necessary to support multi-party add-on conference call requirements. Specify if peripheral hardware equipment, e.g., conference bridge servers, is required.

Avaya Response:

Avaya Communication Manager natively supports conferencing capabilities with up to five other conferees without adding additional hardware or software (no additional servers are required.) Users with multi-line telephones can set up a conference call between conferees either internal or external to the system, and can be local or long distance.

Users can toggle or swap connections to multiple conference parties (alternately placing each called party on soft hold) with the Toggle/Swap button. The caller can still press the Conference button to conference with all the called parties, or can press Transfer to drop his/her own connection, thereby conferencing only the others (called parties). The Selective conference party display and drop (or forced release on the attendant console) feature uses repeated presses of the Conference Display button to cycle through the display of the names and numbers (if available) of all parties on the call. The caller may selectively drop or mute any party on the conference.

The solution also natively supports “meet-me” conferencing, via programming a special Vector Directory Number (VDN), secured via an access code, and to allow up to six parties to join a conference. No additional hardware, software, or additional circuits are required.

With IP endpoints, the conferencing feature requires interaction with the media gateway, so if this feature is activated during a call between two IP endpoints, they shuffle back to the VoIP module., at the conclusion of the and only the two IP parties remain, the IP stations shuffle back to a direct IP to IP connection.




3.4 VoIP Overflow Traffic If available WAN circuits connecting the HQ, RO and all SB facilities are busy, call admission control levels are reached, or QoS levels are not satisfied on-network voice traffic must be able to automatically overflow to PSTN trunk circuits.


Confirm that your proposed communications system supports overflow of voice traffic across VoiceCon locations if WAN links are not available or conditions are not acceptable. Also indicate if overflow traffic can revert back to the WAN if conditions permit.

Avaya Response:

Comply. Avaya Communication Manger supports the bypass of IP trunks if QoS levels are not met. The specific metrics that can be set are Round Trip Time and packet loss. An administrator can set thresholds for the bypass operation and can also set different thresholds for the re-establishment of the normally used trunks. In the case of trunk circuits being busy an administrator can choose to re-route over alternate trunking facilities by specifying a secondary route within the route plan form. All of the bypass functionality works independently of the type of trunk used for bypass (i.e. analog, ISDN).

3.5.0 Port Interface Circuit Cards For each of the following port types, provide a brief description of the proposed port interface circuit card(s) and/or media gateway equipment included with the proposed IPTS to support analog, digital, and IP ports. Include in the descriptions below the number of port interface terminations for each port circuit card, and the number of available gateway channels for each media gateway unit.

Avaya Response:

Comply. Responses included as part of the items described in Sections 3.5.2 thru 3.5.7.

3.5.1 IP Telephones (desktop instrument and PC client softphones, including Attendant Console Position) & IP Audioconferencing Units


Provide a brief description how all IP telephone types are logically and physically supported by the common control call telephony server. If direct call control signaling via the Ethernet LAN/WAN is not supported identify all intermediary carrier, signaling interface and/or media gateway equipment that is required.

Avaya Response:

Comply.

3.5.2 Analog telephones Vendor Response Requirement:

Provide a brief description how analog telephones are logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Avaya Response:

Comply. The methods of supporting analog telephones are described below.




Media Gateways – Analog Stations

Analog stations are supported using the Avaya MM711 Analog Media Module, which provides eight analog telephone or trunk ports. See the following illustration for an example of the MM711.

Avaya MM711 Analog Media Module

The MM711 provides the administrator with the capability to configure any of the eight ports of this analog circuit pack as:

¾ A loop start or a ground start central office trunk

o Loop current 18-60mA

¾ A wink start or a immediate start Analog Direct Inward Dialing (DID) trunk

¾ A 2-wire analog Outgoing CAMA E911 trunk, for connectivity to the PSTN

o MF signaling is supported for CAMA ports

¾ Analog, tip/ring devices such as single-line telephones with or without LED message waiting indication


¾ Type 1 and Type 2 Caller ID


o ALM

o TST

o ACT

o MM716

o ANALOG

o VH0


The MM716 provides 24 analog ports supporting telephones, trunks, modem, and fax. The 24 ports are provided via a 25 pair RJ21X Amphenol connector, which can be connected by an Amphenol cable to a breakout box or punch down block. These ports can be configured as DID trunks with either wink start or immediate start.


The MM716 provides you with the capability to configure any of the 24 ports as:

¾ A wink-start or an immediate-start DID trunk




¾ Analog tip/ring devices such as single-line telephones with or without LED message waiting indication


¾ Three ringer loads, which is the ringer equivalency number for up to 2,000 feet (610 meters) for all 24 ports

¾ Up to 24 simultaneously-ringing ports

¾ Type 1 caller ID and Type 2 caller ID


3.5.3 Facsimile terminal Vendor Response Requirement:

Provide a brief description how facsimile terminals are logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Avaya Response:

Comply. T.38 has emerged as the key standard for ensuring that VoIP capable networks can continue to support user requirements for real-time fax communications, and enables enterprises to support fax transmissions over IP in a multi-vendor environment. T.38 Fax relay is the ITU-T defined standard for Fax relay. It involves H.323/H.245 signaling capabilities exchange for Fax relay. Fax bearer traffic is relayed (encoded/decoded) according to the T.38 specification.

Avaya has incorporated this standard into its VoIP media modules for media gateways so that they will interoperate with non-Avaya systems such as the Clarent or Alcatel Gatekeeper with Cisco and Clarent Gateways. Avaya Communication Manager systems can also use T.38 to transmit Faxes between other Avaya Communication Manager systems and Modular Messaging. Selection of T.38 for fax transmission involves Avaya Communication Manager negotiating with the far end system to ensure it will support T.38.

Avaya Media Gateways communicate with the fax devices using the standard ITU-T T.30 protocol used by all analog fax devices today. Using modem modulation, the sending fax machine sends T.30 protocol and fax image data to the media

gateway, which demodulates the signals and repackages them into T.38 packets. The T.38 protocol provides an ITU-standard mechanism for a gateway/controller to inform another gateway of the desire to change the

media stream from a voice stream to a data stream. Communication Manager provides this signaling to external T.38-enabled gateways to

negotiate the session. The sending gateway then sends the T.38 packets to the receiving T.38-enabled gateway over IP (UDP), which then delivers the packets using the T.30 protocol to the endpoint fax device.

The Avaya implementation of the ITU’s T.38 standard is intended to interoperate with any T.38 compliant product. Remember this means relay mode only. There is a difference between compliance to a specification and certification from Avaya. Avaya has tested interoperability with Cisco version 2 gateways, Clarent gatekeeper, Alcatel gatekeeper, and has certified Multi-Tech.




General Fax/Modem Call Flow for Avaya Communication Manager

Modems and faxes can also be supported as analog endpoints.

3.5.4. Modem Vendor Response Requirement:

Provide a brief description how modem terminals are logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Avaya Response:

Comply. Please refer to item 3.5.3.

3.5.5 Power Failure Transfer Station (PFTS) Vendor Response Requirement:

Provide a brief description how analog telephone instrument Power Failure Transfer Stations (PFTSs) are logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Avaya Response:

Comply. The Avaya Media Gateway uses traditional PFT units in the wall field to support PFTS locations to directly connect a designated analog station and trunk on the system. The PFT units are included in the proposed design. Emergency transfer control provides -48VDC to operate an external emergency transfer panel. Avaya Communication Manager controls the state of the emergency transfer. (Note that, in the past, hardware boards or alarm panels provided a 3-position physical switch to control emergency transfer.)




Avaya Media Gateways use Emergency Transfer (ETR) logic to support power failure transfer, as described below.

The Emergency Transfer logic, can connect the chassis ‘trunk’ port to one of the line ports

When ETR is on, the chassis trunk port 1 is connected to line port 1 and line port 2 is disconnected

When ETR is on, the chassis analog ports are unmanaged by the call controller

¾ The Media Gateway reports to Avaya CM about V7 presence only when ETR is off

¾ ETR will be ON when Power to the Media Gateway is OFF

¾ Inability to contact any of the call controllers

¾ ETR is administratively ON (CLI, SNMP) Severe software fault

¾ ETR will be OFF when ETR is administratively OFF

¾ ETR is administratively AUTO and there is a connection between the Servers and an ACM (providing the Current Loop detector shows there are no ongoing calls)

¾ After reboot, the Media Gateway will wait until the Current loop Detector shows that there are no active calls between the trunk and the line before reporting ‘V7 insertion’ to Avaya CM

¾ ETR Special LED Indication

For emergency power failure transfer at all Media Gateway locations where this feature is desired VoiceCon will need to supply a 48VDC 90 milliamp power supply.

3.5.6 GS/LS CO Trunk Vendor Response Requirement:

Provide a brief description how GS/LS CO trunk circuits are logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Avaya Response:

For the Avaya Media Gateways proposed CO trunk circuits are supported using the Avaya MM711 Media Module, which provides analog trunk and telephone features and functionality. Please refer to item 3.5.2 for a complete description.




3.2.7 DS1/T-1 Carrier Interface Trunk Vendor Response Requirement:

Provide brief description how DS1-based T-1 carrier trunk circuits are logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Avaya Response:

The Avaya MM710 T1/E1 Media Module terminates a T1/E1 connection. The MM710 has a built-in Channel Service Unit (CSU) so that an external CSU is not necessary. See the following figure for an example of the MM710.

Avaya MM710 T1/E1 Media Module

Highlights of the MM710:

¾ Software selectable T1 or E1 operation

¾ An integrated CSU

¾ Both A-law (E1) and µ-law (T1) gain control and echo cancellation ability

¾ D4, ESF, or CEPT framing

¾ ISDN PRI capability (23B + D or 30B + D)

¾ AMI, ZCS, B8ZS (T1) or HDB3 (E1) line coding

¾ Trunk signaling to support US and international CO or tie trunks

¾ Echo cancellation in either direction

¾ Fractional T1 support

¾ An OIC DB 25-pin interface

¾ A Bantam loopback jack that is used for testing of T1 or E1 circuits.

The MM710 supports the universal DS1 that conforms to the ANSI T1.403 1.544 Mbps T1 standard and to the ITU-T G.703 2.048 Mbps E1 standard.

3.2.8 Other Trunk Interfaces VoiceCon may need at some future time additional analog trunk interfaces, specifically Auxiliary, FX, and E&M Tie Line.


Provide a brief description of how additional analog trunk interface requirements can be logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Avaya Response:

All of the Media Gateways support trunks described above via included hardware.

Note: MOH for VoiceCon is provided through analog ports at each location in this design.



Part 1 - Section 4 - Voice Terminal Instruments

4.0.0. Voice Terminal Instruments The proposed communications system must be able to support a mix of analog and IP communications devices. VoiceCon will provide its own analog telephone instruments, fax terminals, and modems.

Avaya Response:

Read and understood

4.1 Regulation Requirements All single- and multi-line IP phones will be manufactured in accordance with Federal Communication Commission hearing aid compatibility technical standards contained in Section 68.316 and the Telecommunication Act of 1996.


Confirm the proposed telephone equipment satisfies these regulation requirements

Avaya Response:

Comply. Avaya has a rich history of creating solutions that help individuals with disabilities participate more fully in life. Our corporate heritage and commitment predate by decades the laws that require such products. For example, our first TTY software for messaging systems was developed more than 12 years ago; our attendant console that provides many of the capabilities of our Universal Access Phone Status software was developed more than 20 years ago.

Many of the engineers who developed these systems are still with Avaya and working on products that demonstrate our continued commitment to the principles of equal access.

4.2 Desktop IP Telephone Instruments VoiceCon has a requirement for several types of desktop IP telephone instruments:

• Economy

• Administrative

• Professional

• Executive


In a separate PPT file attachment provide a slide illustration (graphic or photograph) of the four proposed desktop IP telephone instruments with models identified. Include in the illustration any add-on modules required to satisfy the individual model requirements.

Avaya Response:

Comply. The proposed design includes native IP telephones for all instruments specified in this requirement, except the analog phones. We have included Avaya SoftConsole application licenses for the required attendant positions.




The IP models included are:

Economy Avaya 1603 IP Telephones

Administrative Avaya 9650 IP Telephones

Professional Avaya 9630 IP Telephones

Executive Avaya 9640 IP Telephones

Call Center Avaya 9650 IP Telephones

All of the proposed IP telephones support IEEE 802.3af standard compliant Power over Ethernet (PoE).

Please see Avaya Appendix 3 Slides 8 to 12 for pictures

4.2.1 Economy Desktop IP Telephone Instrument A single line Economy model will be used in common areas. It should have, at minimum, the following design attributes and features/functions:

• 12-key dial pad

• Hold button

• G.711/G.729 voice codecs

• Auto Self Discovery/DHCP

• QoS Support (802.1p/Q, DiffServ)

• Echo Canceller

• IEEE 802.af POE support


Confirm that your proposed Economy model fully satisfies each of the stated requirements and provide a brief description of the instrument (including supported communications protocols).

Avaya Response:

Comply. The proposed solution includes Avaya 1603 IP Telephones. The Avaya 1603 is a cost effective, entry-level IP telephone designed for walkup and everyday user profiles (this telephone is especially practical when used in conjunction with the Avaya IP Softphone).

The following are characteristics of an Avaya 1603 IP telephone:

¾ Each line button includes dual LEDs (red, green) providing explicit status for the user. For a familiar look and feel, these telephones include several fixed feature keys for common telephone tasks including conference, transfer, drop, hold and mute.

¾ The display on all phone models is backlit for easier viewing in all lighting conditions.

¾ A high quality speakerphone

¾ Supports three line appearances/feature keys




Attributes Yes/No 12-key dial pad Yes

Standards-based codec support: G.711, G.726, G.729A/B Yes

Auto Self Discovery/DHCP Yes

Echo Canceller Yes

QoS Support (802.1p/Q, DiffServ) Yes

Hold key Yes

Last Number Redial key Yes

Release key Yes

Message Waiting/Call Ringing indicator(s) Yes

2 Way Speakerphone Yes Speaker/Mute key Yes

Volume Control keys/slide (separate volume levels in the handset, speaker, Yes and ringer) Backlit display – 2.6” diagonal, 2 rows by 16 characters Yes

Optional IEEE PoE 802.3af adapter registers as class 1 device Yes

Please see Avaya Appendix 3 Power Point Slides slide 8 for photos of the 1603.

4.2.2 Administrative Desktop IP Telephone Instrument The Administrative model will be used by station users who have executive management group call answering and coverage responsibilities. It should have, at minimum, the following design attributes and features/functions:

• 12-key dial pad

• Sixteen (16) programmable line/feature keys with soft label/status indicators

• Capable of supporting an optional add-on key module (12 line/feature, minimum) with soft label/status indicators

• G711, G729 and wideband (G.722 or equivalent) voice codecs


• Echo Canceller


• Hold key

• Last Number Redial key

• Release key

• Message Waiting/Call Ringing indicator(s)

• Full Duplex Speakerphone

• Speaker/Mute key

• Volume Control keys/slide




• High resolution, backlit, monochrome grayscale pixel-based, graphical display screen

• Four (4) associated context sensitive soft feature keys

• Cursor/navigator interface for display control

• LDAP access

• Stored Call Data (Last 50 numbers dialed/Last 50 incoming call numbers)

• Integrated Ethernet switch with two (2) RJ-45 connector interface ports for 10/100 Mbps LAN and desktop PC connectivity

• Bluetooth headset interface



Confirm that your proposed Administrative model satisfies each listed requirement by providing a check mark in the following table. Provide a brief description of the proposed telephone instrument, including supported communications protocols and the extent that any requirement is not fully satisfied, e.g., soft feature key substituted for fixed feature key requirement.

Avaya Response:

Comply. The Avaya 9650 IP Telephone exceeds the requirements listed.


The 9650 is designed to meet the needs of an Administrative Assistant or Receptionist for improved call handling and call management; it supports 24 call appearance/Communication Manager features and can be used with the optional SBM24. The Avaya 9650 IP telephone provides 11 one touch buttons for quick access to Communication Manager Features or Call Appearances.


16 programmable line/feature keys with soft label/status indicators Yes

Add-on key module (12 line/feature, soft label/status indicators) option Yes

G711, G729 and wideband voice codecs Yes


Echo Canceller Yes


Hold key Yes


Release key Yes


Full Duplex Speakerphone Yes

Speaker/Mute key Yes

Volume Control keys/slide Yes




Attributes Yes/No High resolution, backlit, monochrome grayscale pixel screen Yes

Four (4) associated context sensitive soft feature keys Yes

Cursor/navigator interface for display control Yes

LDAP access Yes

Stored Call Data (Last 50 numbers dialed/Last 50 incoming call numbers) Yes

Integrated 10/100 Mbps Ethernet switch (2 RJ-45 connector ports) Yes

Bluetooth headset interface Yes

IEEE 802.af POE support Yes

4.2.3 Professional Desktop IP Telephone Instrument The Professional model will be used by VoiceCon managers. It should have, at minimum the following design attributes and features/functions:

• 12 key dial pad

• Six (6) programmable line/feature keys with soft label/status indicators



• Echo Canceller


• Embedded Web services support, e.g., XML

• Hold key


• Release key





• High resolution, backlit, monochrome grayscale pixel-based, graphical display screen with four (4) associated context sensitive soft feature labels ((key, cursor, or navigator control)

• LDAP access


• Integrated Ethernet switch with two (2) RJ-45 connector interface ports for 10/100/1000 Mbps LAN and desktop PC connectivity.

• Bluetooth or USB interface






Confirm that your proposed Professional model satisfies each listed requirement by providing a check mark in the following table. Provide a brief description of the proposed telephone instrument, including supported communications protocols and the extent that any requirement is not fully satisfied, e.g., soft feature key substituted for fixed feature key requirement.

Avaya Response:

Comply. The Avaya 9630 IP Telephone exceeds the requirements listed by adding an optional Avaya ABT35 Bluetooth Headset

Please see Avaya Appendix 3 Power Point Slides slide 9 and 10 for photos of the 9630.





Echo Canceller Yes


Embedded Web services support Yes

Hold key Yes


Release key Yes





High resolution, backlit, monochrome grayscale pixel display Yes



LDAP access Yes


Integrated 10/100/1000 Mbps Ethernet switch (2 RJ-45 connector ports) Yes

Bluetooth or USB interface Yes


A member of the Avaya one-X Deskphone Edition family, the 9630 IP Telephone is specifically designed for the essential telephone user—those for whom the telephone is essential in order for them to complete their job. Sales people, relationship managers, and attorneys are typical examples of the essential users’ profile.




The 9630 provides superior, high fidelity audio, built in "one touch" access to key Avaya Communication Manager mobility features, and a stylish and professional design.

The 9630 IP Telephone features a 3.8 inch (9.65 cm) diagonal monochrome backlit display which has been enhanced with higher resolution (1/4 VGA) versus other available monochrome telephones from Avaya. The 9630 supports up to 24 call appearances / administered feature keys - with six concurrent line appearances visible at any time.

The 9630 has several LED buttons throughout the front of the phone. Six LED line appearance buttons on the side of the display provide explicit status of different line appearances and administered features, while LEDs built into several buttons on the phone such as Mute, Message, and Headset provide an intuitive and simple experience for the everyday end user.

The 9630 supports an optional 24 button expansion module. This provides the essential user with additional call appearances, bridged appearances and administered feature keys including speed dials.

The 9630 requires electricity from either an 802.af Power over Ethernet switch or a local Avaya power supply. The 9630 is a Class 2 PoE device. Additionally, when configured with an SBM24 button expansion module, the 9630 supplies power to the SBM24 - via the module interface on the phone. No separate power supply is required for the SBM24. Even in this configuration, the 9630 plus SBM24 are still PoE class 2.

ABT-35s Bluetooth Headset (Optional)

The optional ABT-35Ss is a comprehensive wireless headset system designed to seamlessly support the communications needs of today’s business traveler. Based on Bluetooth wireless technology, the ABT-35s headset and base unit supports the Avaya desktop phone as well as any Bluetooth compatible cell phone or handheld device. With this single, lightweight headset, users will experience the freedom, convenience and comfort of hands free voice communications whether in the office or on the go.

In the office, the ABT-35S base unit attaches to the headset / handset jack of the Avaya desktop telephone (both Avaya IP and Digital phones are supported). The headset allows for wireless freedom with clear voice quality up to 33 feet (10 meters) from the base. In addition, the headset itself supports alerting and switch hook control of the Avaya desk set – allowing users to answer incoming calls and carry out conversations – even when away from their desks.

Gigabit Adapter (Optional)

The Integrated Gigabit Adapter for one-X Deskphone Edition IP Telephones seamlessly integrates into the bottom of the 9600 series telephones, providing 10/100/1000 Mbps support in a simple, elegant manner. The 10/100/1000 Ethernet switch is capable of supporting the layer 1 and 2 features as configured for the Ethernet switch built into the telephone such as speed and duplex settings, the ability to report the actual speed and duplex when auto-negotiation is used, the ability to disable the secondary interface, VLAN tagging/removal, VLAN separation, and control of the forwarding of frames with group multicast addresses for 802.1X.




9630G IP Telephone (Optional)

The 9630G provides the same features and functionality as the 9630, along with native support for Gigabit Ethernet. The 9630G is ideal for those customers who require 10/100/1,000 Mbps bandwidth to the desktop. And with its built-in Ethernet switch, the 9630G also provides GigE support to a co-located laptop or workstation. User interface simplified and enhanced.

4.2.4 Executive Desktop IP Telephone Instrument The Professional model will be used by VoiceCon’s executive management team. It should have, at minimum the following design attributes and features/functions:

• 12 key dial pad

• Ten (10) programmable line/feature keys with soft label/ status indicators



• Echo Canceller



• Hold key


• Release key





• High resolution, backlit, color pixel-based, graphical display screen with four (4) associated context sensitive soft feature labels (key, cursor, or navigator control)

• LDAP access


• Integrated Ethernet switch and two (2) RJ-45 connector interface ports; 10/100 Mbps connectivity




Confirm that your proposed Professional model satisfies each listed requirement by providing a check mark in the following table. Provide a brief description of the proposed telephone instrument, including supported communications protocols and the extent that any requirement is not fully satisfied, e.g., soft feature key substituted for fixed feature key requirement.




Avaya Response:

Comply, the Avaya 9640 IP terminal meets or exceeds the requirements and same capabilities as the 9630 listed above, the 9640 does have a High resolution, backlit, color pixel-based, graphical display screen as requested.






Echo Canceller Yes



Hold key Yes


Release key Yes





High resolution, backlit, color pixel display Yes



LDAP access Yes


Integrated 10/100/1000 Mbps Ethernet switch (2 RJ-45 connector ports) Yes






4.2.5 ACD/Supervisor IP Telephone Instrument The ACD Agent//Supervisor model will be used by VoiceCon contact center personnel. It should have, at minimum the following design attributes and features/functions:

• 12 key dial pad

• Six (6) programmable line/feature keys with soft label/status indicators



• Echo Canceller



• Hold key


• Release key

• Work Code key

• Supervisor Alert key





• High resolution, backlit, monochrome grayscale pixel-based, graphical display screen with four (4) associated context sensitive soft feature labels ((key, cursor, or navigator control)

• LDAP access


• Integrated Ethernet switch with two (2) RJ-45 connector interface ports for 10/100/1000 Mbps LAN and desktop PC connectivity.


• Recording interface


The ACD model should also allow agents to have access and be able to view, at minimum, the following types of information in the instrument display field:

• Current ACD system and group call volumes;

• Group agent status;

• Number of calls in queue;

• Longest time for call waiting in queue;

• Incoming call ANI/CLID;

• Variety of personal call statistics.





Confirm that your proposed ACD model satisfies each listed requirement by providing a check mark in the following table. Provide a brief description of the proposed telephone instrument, including supported communications protocols and the extent any requirement that is not fully satisfied, e.g., soft feature key substituted for fixed feature key requirement.

Avaya Response:

Comply with explanation. The Avaya 9650 IP Telephone exceeds the requirements listed except for the inclusion of a RIM interface. RIM is mostly done over IP via a Nice or Witness solution especially where Sarbanes-Oxley is concerned. If VoiceCon requires a traditional RIM at the desktop then other features may be lost.


The 9650 is designed to meet the needs of an ACD Agent/Supervisor for improved call handling and call management; it supports 24 call appearance/Communication Manager features and can be used with the optional SBM24. The Avaya 9650 IP telephone provides 11 one touch buttons for quick access to Communication Manager Features or Call Appearances





Echo Canceller Yes



Hold key Yes


Release key Yes

Work Code key Yes

Supervisor Alert key Yes





High resolution, backlit, monochrome grayscale pixel display Yes



LDAP access Yes





Attributes Yes/No Integrated 10/100/1000 Mbps Ethernet switch (2 RJ-45 connector ports) Yes


Recording interface No

802.11af POE Yes

Display Screen Data/Stats Yes/No Current ACD system and group call volumes Yes

Group agent status Yes

Number of calls in queue Yes

Longest time for call waiting in queue Yes

Incoming call ANI/CLID Yes

Variety of personal call statistics Yes

4.2.6 Desktop IP Telephone Instrument Web Services Functionality Vendor Response Requirement:

Provide a brief description of embedded Web-browser functionality for the proposed Professional and Executive IP desktop telephone instrument models. Include the following information in your response:

• Browser protocols (XML, HTML, WAP, et al.)

• Station user interaction (touchscreen and/or keypad control cursor control);

• Screen saver

• Standard applications, such as visual mailbox, personal directory/calendar, web page display, visual and/or audio alerts

Avaya Response:

Comply. The web browser application provided with the presented Avaya 96xx Series IP Telephones is intended to be a relatively simple interface that can be used to access Wireless Markup Language (WML) based web information. VoiceCon may use off-the-shelf WML web authoring tools for intranet web sites. If administered, a URL will be downloaded to the telephone that is used as the “home” page for this application. Navigation is via activation of links on the home page (and on subsequent pages), and via Back, Forward, Home, Refresh and scrolling controls.

The line-based interface and 4-grayscale display provide an interface similar to that of a Wireless Application Protocol (WAP) telephone or a Personal Data Assistant (PDA) such as an early Palm Pilot. For that reason, the Avaya IP Telephones use Wireless Markup Language (WML) to interact with an external web server. The IP telephones convert the WML sent to it by this external server, and pass Softkey and feature button presses back to the server as appropriate.




broker's

Phone Application Suite

The 9600 Series IP telephones have an embedded WML browser within their displays and support the Avaya PUSH API. Current Avaya Phone Application Productivity Pack applications from Citrix such as Broadcast Server and Network Directory are supported on the 9620 and 9630. In addition, other DevConnect partner companies such as Millenigence, CalAmp, IPcelerate and our own Avaya CSI organization have created phone applications available for the 9600 Series telephones.

Some of the examples of an application for the “Push” capability are:

¾ Broadcasting company news

¾ Sending meeting reminders with conference bridge numbers so that users don't have to search for the conference number

¾ Streaming music such as wake-up alarms in hotel rooms

¾ Streaming announcements for academic environments such as college campuses

¾ Sending critical stock news information to the stocktelephone

¾ Broadcasting critical weather alerts to employees

¾ Building intelligent databases to target information to an individual or groups of phones.

Please note – the “Push” application is an optional feature

4.2.7 Desktop Instrument Options and Add-on Modules Vendor Response Requirement:

Provide a brief description of all hardware/software options and/or add-on modules currently available with the proposed Economy, Administrative, Professional, and Executive models. Options/modules may include key modules, display modules, Bluetooth interface, USB interface, Gigabit Ethernet connectors, et al. necessary to satisfy the above telephone model requirements. Indicate the specific models that support the individual option/module.

Avaya Response:

SBM24 Button Expansion Module

The SBM24 Expansion Module is supported on the 9630, 9640 and 9650 telephones. This module provides 24 additional programmable buttons, which may be configured as call appearances, bridged appearances, or feature keys. The SBM24 Expansion Module is powered from the telephone via the module interface port, so the expansion module does not require a separate power supply.

Please see Avaya Appendix 3 Power Point Slides slide 19 for photos of the SMB24 Expansion Module.




Avaya Gigabit Adapter for 9600 Series

¾ PoE and local power, Class 3

¾ Adapter provides investment protection, migrate at your own speed

¾ Integrated Gigabit ports support green fields and infrastructure-in-place customers

¾ Compatible with all 9600 Series IP Phones that have an adapter interface

¾ Requires the use of the Wedge Stand (sold separately)

Please see Avaya Appendix 3 Power Point Slides slide 20 for photos of the SMB24 Expansion Module.

Bluetooth Adapter for 9600 Series

Integrated Bluetooth Adapter for one-X Deskphone Edition IP Telephones seamlessly integrates into the bottom of the 9600 series telephones, providing wireless Bluetooth connectivity to other Bluetooth enabled communications devices - including 3rd party Bluetooth headsets

Alternate color faceplates

Alternate color faceplates are now available for all 9600 series models - including the SBM24 button module. Beyond standard charcoal gray, five alternate colors are available: Red, Blue, White, Yellow, and Silver. The faceplates are sold by phone model, and by color, in packs of 25.

Please see Avaya Appendix 3 Power Point Slides slide 25 for photos of the optional faceplates.

Avaya Phone Application Suite

Optionally, Avaya has adjuncts and third party applications that provide more robust functionality, such as access to the end users outlook calendar, video, broadcasting of a variety of message and screens, as well as other industry specific applications.

The Avaya Phone Application Suite creates a new communication and application paradigm for Avaya IP telephone users. Beyond the hard dollar cost savings associated with IP telephony, with the Avaya Phone Application Suite, the 9600 series IP telephones become multi-function, intelligent endpoints on the IP network - capable of running productivity enhancing applications outside the context of a telephone call.

The initial release of the Avaya Phone Application Suite provides the AG250 Phone Application Gateway server along with Broadcast Server and Design Studio applications. The Broadcast application is supported on the Avaya 9630 9640 and 9650 IP Telephones.




Components

The Avaya Phone Application Suite consists of several components. Broadcast Server is a powerful application allowing for text messages, graphics and audio to be pushed out to Avaya IP phones.

Design Studio is a software tool that allows existing web based applications to be transformed and made accessible right from 9600 series IP phones. The AG250 Phone Application Gateway is a rack mountable server, which integrates with Avaya Communication Manager and allows for applications to be accessed from Avaya IP telephones. With implementation and maintenance support from the qualified engineers within Avaya Global Services, the Phone Application Suite is a complete solution for desktop productivity.

Avaya AG250 Phone Application Gateway

Broadcast Server

The Broadcast Server application leverages a customers IP infrastructure and 9600 Series IP telephones by allowing an administrator to broadcast audio, text, and images to the displays and speakers of the phones. This allows for simple communications to reach a large number of workers instantly. Say for example, the IT department needs to take a server off line momentarily, an administrator could send a broadcast which plays a horn sound through the phone speaker, then displays the notification about the server going off line. Unlike email, the Broadcast Server application is ideal for cutting through all the clutter and gets information communicated quickly and efficiently. Broadcast messages can be composed and delivered on the fly in an ad hoc manner, or can be easily created and scheduled for future delivery. This is useful in scenarios where simple reminder messages are required or helpful.

Text Messaging

The Text messaging application allows workers to both send and receive simple text messages utilizing their IP telephones. Users can pick from pre-created, simple messages such as "running late" or "I'm leaving now", or messages can be composed using the dial pad of the phone and then sent to one or multiple recipients. Messages can only be sent to other IP phones. This application is designed for use between devices behind VoiceCon’s firewall.

Broadcast Server: distribution lists

The Broadcast Server application supports a browser based interface for the administrator to compose and broadcast messages out to Avaya IP telephones. The Broadcast Server supports the use of distribution lists for the sending of messages. Like with email or voicemail, distribution lists make broadcasting of information easier and more efficient.

Text messaging: support for multiple recipients

The Text Messaging application supports the sending of text messages to multiple recipients simultaneously. So for example,

in the case of a person running late to join a meeting, they could quickly compose and send a message for the meeting organizer - going instantly to the organizers desk phone and conference room phone. This enhances the ability for workers to collaborate and keep in touch with each other real time.




Transformation of web applications for use on IP Phones

The Design Studio software tool allows for existing web based (HTML, XML) applications to be transformed for use on Avaya IP telephones. Existing web based applications can be easily leveraged and extended to IP phones for access. Avaya Global Services has available a per day Professional Services offer for custom deployments based on Design Studio.

Enhanced Worker Productivity

The Avaya Phone Application Suite allows workers to enhance their productivity by running value added applications right from the displays of their Avaya 9600 Series IP telephones. Worker productivity increases, the ability to communicate and collaborate with others increases as well.

Total Solution from Avaya

Avaya Global Services is available for up front planning, consultation, implementation and maintenance support. VoiceCon benefits from looking to Avaya - a single, accountable solution provider for the Phone Application Suite, for all support needs.

4.2.8 SIP Compatibility It is not required that the proposed desktop IP telephone instruments be programmed to support SIP standards and specifications, e.g., RFC3261, at time of installation and system cutover. It is desirable, however, that each of the four proposed instrument models be capable of SIP support at later time.


Complete the following table to indicate which of the proposed telephone models are native SIP or can be programmed for SIP support when requested by VoiceCon.

Model Default VoIP Protocol (SIP, H.323, Proprietary, Other)

SIP-capable thru firmware download

Economy H.323 Yes (Basic Sip 4Q 08)

Administrative H.323 Yes

Professional H.323 Yes

Executive H.323 Yes

ACD H.323 Yes

4.2.9 Encryption VoiceCon requires that its Administrative, Professional, Executive, and ACD model telephone instruments support voice media and control signaling encryption using Advanced Encryption Standard (AES) or a comparable security standard


Confirm that this requirement is satisfied for each of the five designated telephone instrument models for both bearer and control transmission signals. Also indicate the type of encryption method deployed.

Avaya Response:

Comply Avaya supports the standards below that include Encryption

¾ RFC 2246 – Transport Layer Security (TLS), which is an IETF standard (RFC 2246) that supersedes Netscape’s’ Secure Socket Layer (SSL) and provides a host-to-host data connections with encryption and certification at the transport layer.




¾ RFC 3711 – Secure Real Time Protocol (SRTP), a media encryption standard defined as a profile of RTP, is supported in Avaya Communication Manager Release 4.0 and 9600-series telephones.

4.3 Teleworker Options VoiceCon may require a fixed desktop and/or a mobile soft client teleworker option at some future date.

4.3.1 Desktop Teleworker VoiceCon may require a desktop teleworker option at some future date. An IP desktop telephone instrument should be comparable in function, capabilities, and attributes to the Professional model IP desktop telephone instrument as described in Section 4.2.3. If required the teleworker instrument should be able to connect to the host communications system via a VPN connection.


Confirm that your proposed communications system can support a teleworker IP desktop telephone instrument that is comparable to the standard level IP telephone instrument model as described above. Provide a brief description of the proposed model, including:

• How the instrument physically and logically connects to the host communications system;

• If a standard Internet connection is supported;

• If a VPN connection requires an external gateway;

• Operational procedures required to log-in and log-out to the host communications system;

• Local power requirements;

• E911 support and calling procedures to the teleworker’s local PSAP.

Avaya Response:

IP Softphone makes it easy to place and receive phone calls from your PC or laptop, making it ideal for working from any location with an internet connection – such as home office, a hotel, coffee shop, or even an airport.

It offers simple Graphical User Interfaces (GUIs) and integrates well with Microsoft desktop applications such as Outlook and Internet Explorer, enabling click to dial and screen pops. There are options to add Instant Messaging and Presence as well as videoconferencing for enhanced collaboration capabilities. IP Softphone links with your desk phone so that calls to you will ring at your desk phone and PC.

Avaya IP Softphone 6.0 offers integration with other communication applications such as Microsoft Outlook Communicator, Lotus Sametime, and Citrix Presentation Server and with Service Pack 1, and can now support the Microsoft Vista Business and Ultimate Editions in a standalone configuration.

IP Softphone connects to the S8500C via the LAN external power is not needed since this is a PC based application. The S8500C is in control of the Softphone as it would for any other IP, analog or digital phone. E911 is controlled via the RedSky E911 application please see responses to 1.1.2 and 1.1.2.1.




The IP Softphone telephone client can be configured for

¾ VoIP (Roadwarrior) mode,

¾ Dual Connect (Telecommuter) mode

¾ Shared Control mode.

VoIP (Roadwarrior) mode is suitable for users who have a single, broadband connection from their home office or remote location. In VoIP mode, the audio and call control signaling are run across the IP network for a "pure voice over IP" configuration, and offers a great amount of flexibility due to the ubiquity of IP networking. Voice communication is generally done via a headset connected to the PC.

Dual Connect (Telecommuter) mode is ideally suited for users working from a remote office with a phone line. With this option, call control signaling is maintained and delivered across the IP network, but the voice is delivered across a phone line to either a public switched telephone network or digital line to help ensure toll-quality voice. This capability can be extended to any touch-tone phone or cellular phone.

Shared Control mode enables the control of Avaya IP telephones or Avaya digital (DCP) telephones.

The Avaya IP telephone configuration enables users to log into and control their Avaya IP telephone from the Avaya IP Softphone. Users can speak and listen through their telephone, but unlike the Multi-Phone configuration, users can make and handle calls from both the Avaya IP Softphone interface and the IP telephone. This feature can help improve productivity by integrating the IP Softphone capabilities and Personal Information Managers, such as Microsoft Outlook, with the IP Telephone.

IP Softphone R6.0 offers three user interfaces: "call bar" view, "picture of phone" view and "generic phone" view. When not integrated with another application, the user can select either "call bar" view or "picture of phone" view. When integrated with Microsoft Office Communicator, Lotus Sametime Connect, or Citrix Presentation Server, the user can access Softphone via those applications, "call bar" view or "generic phone" view.

Video Call Support: The Avaya Video Integrator 2.0 module streamlines video call setup using standard web cams. It also helps integrate desktop video conferencing via Polycom

videoconferencing systems. This software modulegroup automatically identifies whether the far-end is video-enabled and

establishes a video connection. Avaya IP Telephony features such as hold, transfer, forward and cover can be applied to a video call. If desired, the video call can be dropped and the audio path remains active. Avaya Video

Integrator 2.0 works in either Shared Control mode or VoIP (Roadwarrior) mode. When used in Shared Control mode, the Avaya IP or Digital telephone can be used for the audio path.

Avaya Video Integrator 2.0 requires IP Softphone 5.2 or above, a video license, and is available for download from the Avaya Support web site.

Instant Messaging Support: Avaya IP Softphone integrates a flexible IP telephone client with a SIP/SIMPLE-based Instant Messaging (IM) client. The IM client incorporates a contact list of other IP Softphone users and makes both phone and IM presence visible to other users. It is simple to toggle between the Softphone and IM applications.




The IM and presence capabilities require registering with the Avaya SIP Enablement Service (SES) platform, which is available separately. Languages supported include Simplified Chinese, Dutch, Parisian French, German, Italian, Japanese, Korean, Brazilian Portuguese, Latin American Spanish, and Russian in addition to English. Please see www.support.avaya.com for CM requirements and IP Softphone version requirements.

Please see Avaya Appendix 3, PowerPoint Illustrations slides 14 and 15 for photos.

4.3.2 Mobile Teleworker VoiceCon may require a mobile teleworker option at some future date.

At minimum the PC soft phone client should have comparable telephony services capabilities to the Professional model IP telephone instrument model described in Section 4.2.3, including the capability to function and operate as a SIP client with Microsoft Outlook compatibility. Mandatory requirements include: multiple contact directories; LDAP/Active Directory access, detailed call logs (minimum 100 incoming and outgoing calls, respectively); click to dial function; virtual fixed feature and speed dial keys.


Confirm that the proposed softphone solution satisfies the stated requirements and provide a brief product description that includes an illustration/photograph (PPT format, only) that depicts the look and feel of an active call screen display. Avaya Response:

IP Softphone enables you to click and dial from your Microsoft Outlook contact list, your Lotus Notes contact list, an LDAP directory, Web page or any Windows application. The Call History log stores, over the minimum requirement, information on incoming calls, outgoing calls, incoming Instant Messages, and outgoing Instant Messages by automatically generating a call record or one-line summary of each call/instant message. You can add or modify notes in the record, delete the call record, dial the selected number from the record, or send an Instant Message to the associated Instant Messaging address.

Local Phone Directory

A local phone directory can be created on the PC or laptop for dialing from IP Softphone. Place calls from the directory by selecting a contact name and clicking on the telephone icon, or by hitting the Enter button on the keyboard.

Integration with the corporate directory through LDAP can be accomplished as well as Access to the Avaya Communication Manager Directory For easy look up of co-worker access information

Integration of IP Softphone with Microsoft Office Communicator

IP Softphone appears as a second tab in Microsoft Office communicator, allowing use of the softphone in a familiar interface. This enables conversations to easily move from IM to Voice when needed. Users are also provided with an Avaya tab to access advanced Communication Manager features such as voicemail access, send all calls, and more.

Integration of IP Softphone with Lotus Sametime

IP Softphone integrates seamlessly with this existing application allowing use of the Softphone in a familiar interface. This enables conversations to easily move from IM to Voice when needed.




Integration of IP Softphone with Citrix Presentation Server

IP Softphone integrates seamlessly with this existing application allowing use of the Softphone in a familiar interface. This enables a corporation to support IP Softphone on their existing Citrix servers and enable their remote workers to enjoy the benefits of onenumber access and enhanced productivity in Dual Connect (Telecommuter) mode.

Integration of Avaya Instant Messaging (IM) and Presence

IP Softphone users can send an Instant Message to each other in addition to using the phone application, as well as track each other's phone or IM presence. A contact list of users with presence availability can be maintained by each user.

Integration with Desktop Video

Avaya Video Integrator enables you to automatically establish a video call when your desktop and the far-end are video equipped.

Remote Access to Avaya Communication Manager Station Features

The IP Softphone accesses the same features as those available on your office desk phone. IP Softphone R6.0 offers two primary user interfaces. When not integrated, the user can select either call bar view or picture of phone view. Avaya IP Softphone R 6.0 also introduces a new generic phone view. When integrated with Microsoft Office Communicator, Lotus Sametime Connect, or Citrix Presentation Server the user can access Softphone via those applications, call bar view, or generic phone view.

Please also see response to 4.3.2

4.4 Soft Attendant Console Attendant operator console requirements are to be satisfied using a PC client softphone application. The attendant console application should include several distinct display fields, such as: incoming call queue and active caller information; release loop keys; feature/function keys; direct station selection (contact directory)/ busy lamp field; trunk groups; minor/major alarms; and messaging. GUI capabilities must support drag & click operations.

At minimum the following information and data must be available in the softphone screen display:

• Number of calls in queue;

• Call appearance status;

• Calling/called party number/name;

• Trunk ID;

• COS/COR;

• Number of calls waiting;

• Call coverage status;

• Time/date;

• Call duration;




• Text messages;

• System alarm notification


Confirm the proposed attendant position softphone solution satisfies the stated requirements, and provide a brief description of the proposed softphone solution when programmed for attendant console operation. Include in the description technical PC requirements necessary to operate the soft client package. Also provide as an attachment a representative illustration or photograph (PPT format, only) that conveys the look and feel of an active call console display screen. Avaya Response:

Comply. Avaya SoftConsole is the only Avaya IP attendant console offering. IP connectivity is available in both a voice over IP (VoIP)

configuration and a dual connection configuration for toll quality audio. With its ability to search internal and external directories, place a

caller's email address into an email application and display detailed caller information on up to six calls simultaneously, SoftConsole puts

more power and control at an attendant’s fingertips. It supports IP and Digital Communication Protocol (DCP) connectivity to

Avaya communications servers to give our customers a choice of solutions that meets their needs.

The SoftConsole solution improves the attendant's experience by delivering a new and intuitive graphical user interface (GUI), comprehensive setup wizards, email integration and enhanced directory capabilities. The GUI features directory information, Direct Extensions Select/Busy Lamp Fields (DXS/BLF), user-movable tool bars and six call appearances.

Attendants can speed and simplify call handling by arranging the screen to meet their needs and by dialing from the DXS/BLF component, directory lookups, feature buttons, or manually from the keyboard.

The directories are stored in the Avaya MasterDirectory Data Manager application, a robust directory database management program included with the Avaya SoftConsole solution that facilitates database management. This database application serves as an information management tool - importing and consolidating directory information from voice and data systems and exporting it to directory-enabled applications. MasterDirectory application can import/export and transfer data via standards-based protocols, including:

¾ ODBC Open Data Base Connectivity

¾ LDAP Lightweight Directory Access Protocol

¾ FTP File Transfer Protocol

¾ SMTP Simple Mail Transfer Protocol

¾ CSV text delimited files

Using these protocols, MasterDirectory application extracts data from multiple sources, applies filters and business logic to consolidate the data, and populates directory services and databases for use by applications. For example the MasterDirectory application can collect information from multiple Avaya Call Processing servers, consolidate the data with Human Resource databases, and send the processed data to an LDAP directory service used by phone attendant applications, intranet white and yellow pages and other applications.

Please see Avaya Appendix 3, PowerPoint Illustrations slide 16 for photos.




Complete the following table for attendant soft console display requirements: Display Characteristic Yes/No Number of calls in queue Yes

Call appearance status Yes

Calling/called party number/name Yes

Trunk ID Yes

COS/COR Yes

Number of calls waiting Yes

Call coverage status Yes

Time/date Yes

Call duration Yes

Text messages Yes

System alarm notification Yes

Minimum Requirements

¾ IP enabled Avaya Communication Server running Avaya MultiVantage applications for IP connectivity

¾ A personal computer with a Pentium® based processor with a minimum of 400 Megahertz clock speed

◊ 128 MB of RAM available to SoftConsole application (256 MB recommended for IP single connectivity)

◊ CD-ROM drive

◊ Microsoft Windows 2000 with Server or Professional Service Pack 2 or later, Microsoft Windows XP Professional with service pack 1 or later.

◊ 15" color monitor, 17" or larger recommended

◊ Microsoft Internet Explorer 5.5 or higher (for online help)

¾ A COM Port dedicated to SoftConsole. (No other application may vie for this port.)

¾ A sound card (to generate the ringing sounds).

◊ IP Connection:

¾ A sound card that supports full duplex operation.

¾ Headset

¾ Network Interface Card (NIC) for Local Area Network (LAN) connectivity in IP environments.

¾ 30 MB of hard disk space are required for the SoftConsole software, plus additional space for SoftConsole directory information.




4.5 IP Audio Conferencing Unit VoiceCon requires a limited number of desktop audio conferencing units with multidirectional, full duplex speakerphone operation. The unit must be native IP.


Provide a brief description of the proposed IP audio conferencing unit and include as an attachment an illustration or photograph (PPT format, only) of the unit.

Confirm that your proposed communications system solution can support an advanced mobile client GUI. List the make/model of all cellular handsets and PDAs that can work with the proposed advanced mobile client 4.

Avaya Response:

Comply. We have proposed the Avaya 4690 Conferencing Station. The unit is available in two models, one with extended microphones for large conference rooms, and an office model, without microphones for smaller conference rooms. The proposed solution does not include extended microphones. A picture of the Avaya 4690 Conferencing Station is included in Appendix 3, PowerPoint Illustrations.

Some benefits of the 4690 include:

¾ Access to complete Avaya Communication Manager feature set

¾ Emulation of a 4612 IP Telephone to Avaya Communication Manager

¾ Global ready (Icon labels & Class B EMC)

¾ Backlit, 4 Line text-based display (Eurofont only)

¾ 4 fixed keys: On/Off hook, Redial, Mute, Volume Up/Down

¾ 3 context sensitive soft keys below display

¾ Local phone features via 4 menu keys

¾ Power supply included

4.6 Mobile Extension VoiceCon requires that the proposed IPTS support mobile extensions for a number of its station users.

4.6.1 Basic Mobile Cellular Extension Option VoiceCon requires that the proposed communications system solution support a mobile cellular extension option.

• The option should be capable of working with almost any cellular carrier network and supported mobile handsets/PDAs;

• The mobile handset/PDA must be able to receive incoming calls directed to the station user’s primary system directory number, and calling party information should be displayed at the mobile handset;

• Calls placed from the mobile handset/PDA to other communications system subscribers must appear to look like calls from the station user’s primary desktop voice terminal, including calling party name/ID display;

4 Removed by Allan during review call.




• IPTS system subscribers must be able to program incoming calls to ring simultaneously or sequentially at the desktop instrument and mobile handset/PDA as required;

• Basic IPTS telephony features that should be supported in mobile extension mode, including Hold, Transfer, Conference, and Forward to IPTS voice mail system on no-answer;

• Call detail records must be collected and stored for all mobile extension calls.

At time of system installation VoiceCon requires 100 mobile cellular extension user licenses. Include this requirement in your pricing proposal.


Confirm that your proposed communications system solution supports mobile cellular extension capabilities as listed above, and is included in the system configuration and pricing proposals. Include a brief description of any hardware/software requirements, including peripheral application servers, necessary to support the option and provide a list of standard feature/function capabilities.

Avaya Response:

Comply. The powerful Extension to Cellular feature of Avaya Communication Manager can connect callers to employees wherever they are with just one call. Avaya Extension to Cellular delivers one-number access by enabling calls bound for an employee’s business number to ring simultaneously on both their office phone and up to four mobile (or wire line) phones. While on a call, users can access advanced features such as transfer and

conference. Extension to Cellular also offers significant cost savings on international calls as well as cost tracking information. Voice Mail is provided via Avaya Modular Messaging so the end user always know where to check for business messages.

Included with Extension to Cellular is the Avaya one-X™ Mobile application for an enhanced user experience.

Access to advanced features can be greatly simplified with the Avaya one-X™ Mobile Edition. The right to use this application is included with the Extension to Cellular license and provides users with simple menu-based access to advanced Communication Manager features from their cell phones including the call detail record for all calls made via the one-X Mobile Edition equipped cell phone.

Extension to Cellular offers dramatically enhanced productivity with one-number access. Extension to Cellular transparently

bridges calls received by the Avaya Communication Manager server to any mobile phone, regardless of location or wireless service provider. Bridging offers many benefits over simple call forwarding, including the ability for the user to seamlessly hand off calls between mobile and desk phone, and access to advanced PBX features (like conferencing and transfer) midcall. Extension to Cellular can be activated or deactivated from an Avaya desktop phone or remotely from the user's mobile phone. In addition, many Avaya Communication Manager features can be accessed while on an extended call. Users can transfer or conference calls and answer calls on multiple lines with their personal mobile phone, improving their productivity while outside of the office.




Extension to Cellular also offers remarkable cost reduction benefits. For employees with significant international calling expenses, Extension to Cellular users can manually route an international mobile call through an Avaya Communication Manager server. This replaces an international cellular call with a mobile-to-PBX call and then PBX-to-international call via existing corporate infrastructure. Extension to Cellular also offers improved cellular cost tracking with call detail recording capability.

One-button Access from the Deskphone. Extension to Cellular users can activate, deactivate, and suspend Extension to Cellular service by using an administered Extension to Cellular feature status button on their desk phones. The Extension to Cellular button remains lit when service is enabled, is off when service is disabled, and flashes when service is suspended through the optional timer. The Extension to Cellular feature status button is available on telephones that support administrable feature buttons. In addition, Extension to Cellular can be turned on and off from the user's mobile phone.

Access to Avaya Communication Manager Station features

Users can now access 28 different Avaya Communication Manager features from their bridged mobile phone. These features include conference and transfer, call forward, call park, send all calls, malicious call trace, active/held/idle line pickup, and whisper page.

Access to advanced features is greatly simplified with Avaya one-X Mobile Edition. The right to use this software is included with the Extension to Cellular license and provides the user with a simple, elegant GUI on their cell phone with menu-based access to advanced Communication Manager features.

For users whose cell phone does not support Avaya one-X Mobile, access to these features can be provided by programming speed dials for one-touch access.

4.6.2 Advanced Mobile Client To optimize the advantages of deploying a mobile cellular extension option VoiceCon also requires that the proposed communications system solution support an advance mobile client option. The mobile client should include a user friendly GUI to facilitate and enhance mobile handset telephony service features and functions. At minimum the mobile client GUI should display: multi-line appearances; fixed and programmable feature keys; contact directories; call logs; incoming call information; active call information.

At time of system installation VoiceCon requires 50 advanced mobile client user licenses. Include this requirement in your pricing proposal.


Confirm that your proposed communications system solution supports an advanced mobile client option, and that is included in the system configuration and pricing proposals. Include a brief description the software client and provide as an attachment one or two graphical illustrations (PPT format, only) that are representative of the GUI screen display.

Avaya Response:

Comply, Avaya one-X™ Mobile for Symbian and Windows Mobile 5 transforms a user's mobile phone into an extension of their office desk phone. This is an entitlement with the Extension to Cellular Solution. Avaya one-X Mobile offers a GUI to the user to enhance the

capabilities of Extension to Cellular. No servers are required, except for Avaya one-X Mobile for Symbian Dual Mode which requires a SIP server.




Avaya one-X Mobile for Symbian and Avaya one-X Mobile for Symbian Dual Mode are available today and support many Nokia devices including the popular E-series. Avaya one-X Mobile for Windows Mobile 5 is also available and offers support on a variety of devices including HP iPAQ, Palm Treo and more. Avaya one-X Mobile supports product from every Service Provider in the United States.

This enables users to have one number, one voicemail, dial-by-PBX, and advanced PBX feature access via an integrated GUI on the mobile device. Avaya one-X Mobile for Symbian Dual Mode offers all of these features in addition to the ability to talk over cellular or WLAN and seamlessly transfer calls between the two.

Benefits to VoiceCon:

¾ One Number - Calls initiated by the mobile device will dial-by PBX and display the caller ID of the desk phone. Calls made to the desk phone extension will ring both the desk phone and the mobile device. The user can easily enable or disable both of these features from the mobile device.

¾ One Voicemail - Unanswered calls will be picked up by the Enterprise voicemail system, eliminating the need to check multiple voicemail systems throughout the day.

¾ Avaya One-X Mobile Dual Mode - This enables the user to connect over cellular or WLAN and seamlessly transfer calls between them. In both modes the user has access to all the normal one-X Mobile feature set. WLAN connectivity offers two main benefits:

¾ Cost Savings - When using a mobile device while on premise,call delivery path is WLAN so service provider minutes are not used.

¾ Increased Connectivity - Users can remain reachable atlocations on premise without cellular service simply by installing WLAN access points.

¾ Handoff Calls between Mobile Device and Deskphone - All calls made via, or received via, Avaya one-X Mobile can be seamlessly transferred between the desk phone and mobile device. An employee arriving at his or her desk while on a mobile call can pick up their desk phone to continue the conversation. While on a desk phone call, the user can become mobile by extending the call to their mobile device. In both cases, the handoff is seamless and is completely transparent to the other party.

¾ Advanced PBX Feature Access - While on a call, the user has access to features such as conference, transfer, call park, and many more that IT selects. These are all accessed via a simple GUI.

¾ Savings on International Calls - Direct dialed international calls from a mobile device can be very expensive. With Avaya one-X Mobile the user can leverage your existing corporate infrastructure from their mobile device. The user would dial the international number on their mobile device, but the international call would be dialed via the Avaya PBX. The mobile device is only making a local call to the PBX, with the PBX dialing the international number.




¾ Business Continuity - With Avaya one-X Mobile, users only need to know each others normal desk phone extensions to remain connected. In the event that employees are unexpectedly unable to go to the office, everyone can remain connected and reachable.

¾ One Device - The user can control whether calls are dialed via-PBX, for workrelated calls, or dialed direct from the cellular device for personal calls, so users will not need multiple devices. Calls which are dialed via-the-PBX are made as though from the user’s desk phone. International calls will enjoy significant savings. Existing infrastructure, such as call recording, will remain useable even for mobile calls. Call detail recording can be used to distinguish mobile calls from desk calls for auditing of cell phone bills.

Avaya one-X™ Mobile is a software client family that enables easy access to the features of Avaya Communication Manager, via an integrated GUI on the mobile device.

4.7 Other IP Telephone Instruments Include as an attachment a graphical illustration (PPT format, only) of IP telephone instrument models and add-on options not included as part of the proposed required system configuration.

Avaya Response:

Comply optional IP telephones that are available include

9610 Walkup Phone 9620 IP Phone

1608 IP Phone 1616 IP Phone

4602 IP Phone 4610 IP Phone

4621SW IP Phone 4622SW IP Phone

4625 IP Phone 16CC IP Call Center Phone

Avaya SIP Softphone

Please see Avaya Appendix 3, PowerPoint Illustrations slides 23 through 35 for photos.



Part 1 - Section 5 - Call Processing Features

5.0 Call Processing Features The proposed communications system should have a robust list of call processing features supporting station user, attendant, and system operations.

Avaya Response:

Comply. Avaya Communication Manager includes more than 700 features. Descriptions of the most widely used features are included in Appendix 1, Avaya Communication Manager Feature Descriptions.

5.1 Station User Features It is required that the proposed communications system support the following list of station user features. Definitions for most listed features may be found in PBX Systems for IP Telephony (2002), written by Allan Sulkin and published by McGraw-Hill Professional.

Table 9 Station User Features Feature Supported ADD-ON CONFERENCE (6 party or more) Yes

AUTOMATIC CALLBACK Yes

AUTOMATIC INTERCOM Yes

BRIDGED CALL APPEARANCE Yes

CALLBACK LAST INTERNAL CALLER Yes

CALL COVERAGE (PROGRAMMED) Yes

INTERNAL & EXTERNAL CALL PROGRAMMING

Yes

TIME OF DAY/DAY OF WEEK CALL PROGRAMMING

Yes

ANI/DNIS/CLID CALL PROGRAMMING Yes

INTERNAL CALLER ID PROGRAMMING

Yes

CALL FORWARDING - ALL CALLS Yes

CALL FORWARDING - BUSY/DON'T ANSWER

Yes

CALL FORWARDING - FOLLOW-ME Yes

CALL FORWARDING - OFF-PREMISES Yes

CALL FORWARDING: RINGING Yes

CALL HOLD Yes

CALL PARK Yes

CALL PICKUP - INDIVIDUAL Yes

CALL PICKUP - GROUP Yes

CALL TRANSFER Yes

CALL WAITING Yes

CONSECUTIVE SPEED DIALING Yes

CONSULTATION HOLD Yes

CUSTOMER STATION REARRANGEMENT

Yes




Feature Supported DIAL BY NAME Yes

DISCRETE CALL OBSERVING Yes

DISTINCTIVE RINGING Yes

DO NOT DISTURB Yes

ELAPSED CALL TIMER Yes

EMERGENCY ACCESS TO ATTENDANT Yes

EXECUTIVE ACCESS OVERRIDE Yes

EXECUTIVE BUSY OVERRIDE Yes

FACILITY BUSY INDICATION Yes

GROUP LISTENING Yes

HANDS-FREE DIALING Yes

HANDS-FREE ANSWER INTERCOM Yes

HELP INFORMATION ACCESS Yes

HOT LINE Yes

INCOMING CALL DISPLAY Yes

INDIVIDUAL ATTENDANT ACCESS Yes

INTERCOM DIAL Yes

LAST NUMBER REDIALED Yes

LINE LOCKOUT Yes

LOUDSPEAKER PAGING ACCESS Yes

MALICIOUS CALL TRACE Yes

MANUAL INTERCOM Yes

MANUAL ORIGINATING LINE SERVICE Yes

USER CONTROLLED MEET ME CONFERENCING (6-Party or more)

Yes

MESSAGE WAITING ACTIVATION Yes

MULTI-PARTY ASSISTED CONFERENCE w/SELECTIVE CALL DROP

Yes

MUSIC ON HOLD Yes

OFF-HOOK ALARM Yes

PADLOCK Yes

PAGING/CODE CALL ACCESS Yes

PERSONAL CO LINE (PRIVATE LINE) Yes

PERSONAL SPEED DIALING Yes

PERSONALIZED RINGING Yes

PRIORITY CALLING Yes

PRIVACY - ATTENDANT LOCKOUT Yes

PRIVACY - MANUAL EXCLUSION Yes

RECALL SIGNALING Yes

RINGER CUT-OFF Yes




Feature Supported RINGING TONE CONTROL Yes

SAVE AND REDIAL Yes

SECONDARY EXTENSION FEATURE ACTIVATION

Yes

SEND ALL CALLS Yes

SILENT MONITORING Yes

STEP CALL Note 1

STORE/REDIAL Yes

SUPERVISOR/ASSISTANT CALLING Yes

SUPERVISOR/ASSISTANT SPEED DIAL Yes

TEXT MESSAGES Yes

TIMED QUEUE Yes

TRUNK FLASH Yes

TRUNK-TO-TRUNK CONNECTIONS Yes

WHISPER PAGE Yes

Note 1: While we support this functionality on an attendant basis, it is not supported on an individual user basis.


Confirm that the proposed IPTS supports each of the above listed station user features. Identify any and all features that are not included as part of the standard call processing software generic package. Also identify optional hardware/software, e.g., CTI application server, to satisfy a listed feature, because it is not included as part of the standard generic software package.

Avaya Response:

We comply to all of the features listed above, as described in the book, PBX Systems for IP Telephony (2002), written by Allan Sulkin and published by McGraw-Hill Professional, except to the STEP CALL feature. For this feature, we partially comply, as explained below.

Step Call

SULKIN DEFINITION: Step Call allows a station user or attendant, after dialing a busy station, to dial an idle station by simply dialing an additional digit. The feature can be implemented only if the dialed digits of the first dialed number and the second number are identical, except for the last digit.

Partially comply. While we support this functionality on an attendant basis, it is not supported on an individual user basis. If the functionality is needed for designated users in VoiceCon’s enterprise, those users can be given the function on an individual basis.

It is important to note that all of the features listed above, as well as many others listed in Appendix 1, Avaya Communication Manager Feature Description, are standard and supported on the current release of the software. Appendix 2 contains a list of Avaya Communications Manager Standard/Optional Features.




5.1.1 Additional Station User Features Vendor Response Requirement

Provide as an attachment a listing of proposed standard generic software station user features that are not included in Table 9 that VoiceCon may find of use and benefit.

Avaya Response:

Please refer to Appendix 2 - Avaya Communications Manager Standard/Optional Features for additional station user features.

5.2 Attendant Operator Features It is required that the proposed communications system support the following list of attendant operator features. Definitions for most listed features may be found in PBX Systems for IP Telephony (2002), written by Allan Sulkin and published by McGraw-Hill Professional.

Table 10 Attendant Operator Features Attendant Features Supported AUTO-MANUAL SPLITTING Yes AUTO-START/DON'T SPLIT Yes BACK-UP ALERTING Yes BUSY VERIFICATION OF TERMINALS/TRUNKS

Yes

CALL WAITING Yes CAMP-ON Yes CONFERENCE Yes CONTROL OF TRUNK GROUP ACCESS

Yes

DELAY ANNOUNCEMENT Yes DIRECT STATION SELECTION w/BLF Yes DIRECT TRUNK GROUP SELECTION Yes DISPLAY Yes INTERCEPT TREATMENT Yes INTERPOSITION CALL & TRANSFER Yes INTRUSION (BARGE-IN) Yes OVERFLOW Yes OVERRIDE OF DIVERSION FEATURES

Yes

PAGING/CODE CALL ACCESS Yes PRIORITY QUEUE Yes RECALL Yes RELEASE LOOP OPERATION Yes SERIAL OPERATION Yes STRAIGHT FORWARD OUTWARD COMPLETION

Yes

THROUGH DIALING Yes TRUNK-TO-TRUNK TRANSFER Yes




Attendant Features Supported TRUNK GROUP BUSY/WARNING INDICATOR

Yes

TRUNK ID Yes


Confirm that the proposed IPTS supports each of the above listed attendant features. Identify any and all features that are not included as part of the standard call processing software generic package. Also identify optional hardware/software, e.g., CTI application server, to satisfy a listed feature, because it is not included as part of the standard generic software package.

Avaya Response:

Comply. All attendant features listed above are supported by Avaya Communication Manager, no additional hardware or software other than the proposed Avaya Softconsole. Appendix 1, Avaya Communication Manager Feature Descriptions, provides additional information.

The proposed Avaya SoftConsole requires a PC for each attendant that is provided by VoiceCon. The specifications are provided in response 4.4 of this proposal.

5.2.1 Additional Attendant Operator Features Vendor Response Requirement

Provide as an attachment a listing of proposed standard generic software attendant operator features that are not included in Table 10 that VoiceCon may find of use and benefit.

Avaya Response:

Please refer to Appendix 2 - Avaya Communications Manager Standard/Optional Features for additional attendant operator features.

5.3 System Features It is required that the proposed communications system support the following list of system features. Definitions for most listed features may be found in PBX Systems for IP Telephony (2002), written by Allan Sulkin and published by McGraw-Hill Professional.

Table 11 System Features System Features Supported ACCOUNT CODES Yes

ADMINISTERED CONNECTIONS Yes

ANSWER DETECTION Yes

AUTHORIZATION CODES Yes

AUTOMATED ATTTENDANT Yes

AUTOMATIC CALL DISTRIBUTION Yes

AUTOMATIC ALTERNATE ROUTING Yes

AUTOMATIC CAMP-ON Yes

AUTOMATIC CIRCUIT ASSURANCE Yes

AUTOMATIC NUMBER ID Yes

AUTOMATIC RECALL Yes

AUTOMATIC ROUTE SELECTION - Yes




System Features Supported BASIC AUTOMATIC TRANSMISSION MEASUREMENT SYSTEM

Yes

CALL-BY-CALL SERVICE SELECTION

Yes

CALL DETAIL RECORDING Yes

CALL LOG Yes

CENTRALIZED ATTENDANT SERVICE

Yes

CLASSES OF RESTRICTION (SPECIFY #)

Yes – 996

CLASSES OF SERVICE (SPECIFY #) Yes – 16

CODE CALLING ACCESS Yes

CONTROLLED PRIVATE CALLS Yes

DELAYED RINGING Yes

DIAL PLAN Yes

DIALED NUMBER ID SERVICE Yes

DIRECT DEPARTMENT CALLING Yes

DIRECT INWARD DIALING Yes

DID CALL WAITING Yes

DIRECT INWARD SYSTEM ACCESS Yes

DIRECT INWARD TERMINATION Yes

DIRECT OUTWARD DIALING Yes

E-911 SERVICE SUPPORT Yes

EXTENDED TRUNK ACCESS Yes

FACILITY RESTRICTION LEVELS Yes

FACILITY TEST CALLS Yes

FIND ME- FOLLOW ME Yes

FORCED ENTRY ACCOUNT CODES Yes

HOTELING (/PERSONAL ROAMING) Yes

HOUSE PHONE Yes

HUNTING Yes

INTEGRATED SYSTEM DIRECTORY Yes

LEAST COST ROUTING (Tariffbased, TOD/DOW)

Yes

MULTIPLE LISTED DIRECTORY NUMBERS

Yes

MUSIC ON HOLD Yes

NIGHT SERVICE –FIXED Yes

NIGHT SERVICE -PROGRAMMABLE

Yes

OFF-HOOK ALARM Yes




System Features Supported OFF-PREMISES STATION (OPX) Yes

OPEN SYSTEM SPEED DIAL Yes

PASSWORD AGING Yes

POWER FAILURE TRANSFER STATION

Yes

RECENT CHANGE HISTORY Yes

RESTRICTION FEATURES: Yes

CONTROLLED Yes

FULLY RESTRICTED Yes

INWARD/OUTWARD Yes

MISCELLANEOUS TERMINAL Yes

MISCELLANEOUS TRUNK Yes

TOLL/CODE Yes

TRUNK Yes

VOICE TERMINAL (IN/OUT) Yes

ROUTE ADVANCE Yes

SECURITY VIOLATION NOTIFICATION

Yes

SHARED TENANT SERVICE Yes

SNMP SUPPORT Yes

SYSTEM SPEED DIAL Yes

SYSTEM STATUS REPORT Yes

TIME OF DAY ROUTING Yes

TIMED REMINDER Yes

TRUNK ANSWER ANY STATION Yes

TRUNK CALLBACK QUEUING Yes

UNIFORM CALL DISTRIBUTION Yes

UNIFORM DIAL PLAN Yes

VIRTUAL EXTENSION Yes

VOICE MESSAGE SYSTEM INTERFACE

Yes


Confirm that the proposed IPTS supports each of the above listed system features. Identify any and all features that are not included as part of the standard call processing software generic package. Also identify optional hardware/software, e.g., CTI application server, to satisfy a listed feature, because it is not included as part of the standard generic software package.

Avaya Response:

Comply. All features listed above are supported as standard on the Avaya Communications Manager, Release 5.0, which provides over 700 features, too many to list below.




Additional detail on the features listed above is provided in Appendix 1, Avaya Communication Manager Feature Descriptions. In addition, we included a Standard and Optional Feature Guide in Appendix 2.

5.3.1 Additional System Features Vendor Response Requirement

Provide as an attachment a listing of proposed standard generic software system features that are not included in Table 11 that VoiceCon may find of use and benefit.

Avaya Response:

Please refer to Appendix 2 - Avaya Communications Manager Standard/Optional Features for additional system features.



Part 1 - Section 6 - Systems Management

6.0.0 Systems Management The proposed communications system must be administered, monitored, and maintained through operations organized into five functional areas: Fault, Configuration, Accounting, Performance and Security. All of the systems and devices in your proposed solution should attempt to provide comprehensive operations in each area.

Operations for each area must be accessible through one interface regardless of the underlying system or device being managed. If a proxy server is used for intermediate operations, there must be at most one central database for each functional area. Systems or devices may be accessed individually if no proxy server is used.

EXCEPTION: Optional call center solutions may provide its own set of FCAPS management operations separate from the general enterprise communications solution.

Any supplied management applications must support decentralized access from any distributed PC client across the HQ LAN/WAN infrastructure and remote dial-up PC clients. It is also desirable for the applications to support a browser based user interface for intensive remote operations.

Any supplied management applications may integrate information from the five functional areas at the presentation level.


Confirm and verify that each functional area required to manage the proposed IPTS network is supported by a single, centrally located proxy server or, alternatively, each system or device supports a single API for a given functional area. Provide a brief description of the proposed management system, including its major hardware and software components. Specify if the proposed systems management server and software is available as a bundled offering, only, or if VoiceCon is responsible for providing its own server hardware to operate the software. If third party technology is used, please indicate which components are managing your solution in a vendor agnostic fashion.

Avaya Response:

Comply. The proposed solution includes the Integrated Management Standard Edition This powerful management application will allow VoiceCon’s System Administrator to centrally manage the proposed Server, Media Gateways, Messaging Server, and optional Wireless LAN Access Points from the same interface. The Avaya Integrated Management applications include the tools that enable VoiceCon to:

¾ Configure, monitor, update, and optimize the performance of Avaya media servers, gateways, and endpoints

¾ Monitor voice over IP traffic

¾ Manage Quality of Service (QoS) policies

¾ Optimize performance

Avaya Integrated Management provides a complete solution for system administration, network management, and provisioning of converged networks.

The Avaya Integrated Management 5.0 applications now run on additional platforms and operating systems.




These include:

¾ Windows 2000/2003

¾ MS Vista support for Windows Applications

¾ VMWare for server-based applications

¾ Internet Explorer 7 for all applications that support internet browser

¾ Java Runtime Environment (JRE) 1.5.11

The Communication Manager Standard Edition license includes Avaya Site Administration, Voice Announcement Manager, VoIP Monitoring Manager and Converged Network Analyzer. Please see responses throughout section 6 for descriptions.

Hardware and Software Requirements:

Windows 2000/2003/Vista Server Hardware

Component Recommended Comments Processor 3.0 GHz (Pentium 4

or AMD Opteron) The Server or the Desktop PC must run Windows 2000/2003/Vista server software. Windows 2000 Server is supported only for upgrades from the prior release

Hard Drive 40 GB Hard Drive Network Connectivity 10/100 Network Card RAM 2 GB RAM Additional RAM is recommended

for better performance Modem or SIG/SSG 56K Modem Needed for remote access by

Avaya Services CD-ROM CD-ROM Needed for installation Extra software Anti-Virus; pcAnywhere

version 10 pcAnywhere is needed for remote access by Avaya Services

Web Browser Internet Explorer 6.0 w/SP1or 7.0

Needed for access to Integrated Home Page and Web based clients

Client PC Hardware and Software

Component Recommended Comments Processor 600MHZ Pentium III Windows 2000/2003/ XP Professional/ Vista Integrated Management

Suite will not allow the client software to load to Windows 98 or NT.

Web Browser Internet Explorer 6.0 or later If the client wants to connect to HP OpenView from the client PC then NetMeeting or pcAnywhere version 10 is needed.

Memory 512MB CD-ROM CD-ROM




Component Recommended Comments Hard Drive 2.5GB Needed if the following

applications are downloaded to the client PC. Avaya Site Administration Avaya Voice Announcement Manager Avaya VoIP Monitoring Manager

6.0.1 System/Port Capacity Vendor Response Requirement

Identify the maximum number of independent IPTS communications systems that can be supported by the proposed systems management server, and the maximum number of user ports that can be passively and actively supported.

Avaya Response:

Comply. The Administration offer is a management solution that provides single or multilicense permission. This will provide VoiceCon with the tools needed to manage a small network of voice systems. 1,000 stand alone devices 4,000 objects Devices include media gateways, data switches/hubs/routers, wireless access points – that are monitored and managed. Objects are defined as items that are represented in the database. These include data switches/hubs/routers, voice switches, IP phones, etc….

Our Administration offer gives VoiceCon a license to install one copy each of Avaya Site Administration and Avaya Voice Announcement Manager.

6.0.2 Terminal Capacity Vendor Response Requirement

Identify the maximum number of configurable and active PC client terminals that can be configured as part of the proposed management server system.

Avaya Response:

The proposed solution includes five user licenses for the Avaya VoIP Monitoring Manager and Converged Network Manager, with one license each for Avaya Site Administration and Voice announcement Manager.




6.0.3 Support for Open Standards The proposed management system should provide support for open protocols, such as LDAP and SNMP. The proposed management system should use open encoding schemes, such as XML and HTML.


Briefly discuss the open standards included in your proposed management system that supports administration, operations and maintenance services. Indicate if any protocols or encoding schemes are de facto standards or are being implemented publicly by other vendors.

Avaya Response:

Avaya VoIP Monitoring Manager is a open standards QoS monitoring and feedback, Windows 2000/2003/Vista software tool that allows the customer to visualize the real-time operation of VoIP systems. The tool provides information on QoS parameters related to VoIP quality.

Avaya VoIP Monitoring Manager provides the ability to view QoS related information through a client GUI application from the customer’s LAN or via remote access. Avaya VoIP Monitoring Manager is also capable of being configured to generate traps, associated with VoIP QoS, sent to any NMS. Avaya VoIP Monitoring Manager can receive RTP Control Protocol (RTCP) packets from the following entities: Avaya IP telephones, Avaya IP soft phones, VoIP engines (on media gateways) and prowler boards. The RTCP data for current calls is published in the Real-time Transport Protocol (RTP) Message Information Base (MIB) via the SNMP agent running on the server. The historical RTCP data is published in the VMON MIB. Both forms of RTCP data can be viewable by either field support, a system administrator or services personnel.

¾ Security enhancements

¾ Authentication with AAA/LDAP

◊ AP-MON, providing performance and traffic analysis

¾ New Hardware Support

◊ Support for the G450 Media Gateway

◊ Support for the S8300C

¾ SystemView – logical view of VoIP device hierarchy

◊ Reflects Avaya Communication Manager servers, LSPs, CLANs, Media Gateways, and IP Phones

The proposed Avaya SoftConsole supports LDAP compliant directories which are managed through VoiceCon’s LDAP application directly.

The S8X00 series has an embedded SNMP native agent, which can send traps to any network management application such as the Network Management Console. Using the SNMP Native Agent VoiceCon can elect to send traps to 5 locations one of which could be a Network Management Console (NMS), they can also download the traps from www.support.avaya.com

The proposed solution runs on a Customer provided Windows 2000/20003 server or on VMware. The VoIP Monitoring Manager also ships with a Microsoft SQL Server 2005 Express Edition which supports maximum database size of 4GB. The customer also has the option of installing their own Windows 2000/2005 SQL database for larger storage capacity and allows the customer to perform SQL queries.




6.0.4 Security Features Unauthorized access to the communications system is a major concern. The ability to detect security problems is desirable beyond mechanisms to prevent security problems.


Briefly describe the security features that are embedded in the proposed management system to prevent unauthorized access and operation. Specify if media encryption is used for command signaling transmissions. What, if any, Denial of Service (DoS) and user authentication mechanisms are supported for the systems management application?

Avaya Response:

All access to the S8X00 Series Communication system requires a valid login on Communication Manager for each user to be able to access.

All products offered through the Integrated Management Suite follow all United States guidelines for encryption algorithms that can be exported

Authentication with the VoiceCon’s AAA/LDAP authentication makes for even more secure access.

6.0.5 User Interface & Tools The management system should be operated using by GUI tools, formatted screens, pull down menus, valid entry choices, templates, batch processing & transactions scheduling, and database import/export. In general you should support a user interface set for each functional area: fault, Configuration, Performance and Security. The constituent users of each of these areas are distinct and your interface for each should optimize the experience for that constituent group. Management applications my integrate information from several management areas to enhance one functional area being managed.

Avaya Response:

Comply. The proposed Avaya Servers and Gateways include embedded native agents. The Modular Messaging application is managed with a web-based device manager, as well as Avaya Site Administration. Our offer includes Voice Announcement Manager, VoIP Monitoring Manager, and Converged Network Manager which support the capabilities listed above. The following optional applications enable you to administer site moves, perform adds and changes, configure distributed networks and campus environments, and manage performance in converged networks.




Avaya Enterprise Network Management Web Pages




Avaya Site Administration (ASA)

Avaya Site Administration (ASA) supports individual Avaya Media Servers and Avaya messaging platforms. ASA is a PC-based Windows (2000/XP Professional/Vista) tool that presents a graphical user interface along with special task wizards that simplify basic administration functions by reducing repetitive tasks for moves/adds/changes, finding extensions, monitoring, and a variety of other functions. A graphical tool provides access to more advanced administrative tasks. Not sure what this last sentence is saying are you talking about MSA?

Using ASA, an administrator can access the switch for administration and maintenance purposes; this can be done remotely from any point on the network, including offsite assuming VPN access is in place so the administrator can log into the switch. A secure connection to the voice and messaging systems can be made through SSH or LDAP authentication. ASA allows the administrator to setup, configure, and troubleshoot any station on the network from the centralized administration platform.

Below are some features of ASA:

¾ Administration Wizards including Add/Change/Remove User and Change User Extension

¾ Terminal emulation and GEDI (Graphically Enhanced DEFINITY Interface) for access to Communication Manager for other administrative operations

¾ Templates for adding new objects

¾ Native Name Entry for Station Object

¾ Import/Export of administration fields for select objects

¾ Report generation

¾ Schedule tasks to run at a later time

¾ Button Label Printing

¾ Basic Fault & Performance capabilities

¾ Email Notification (for Monitoring and Report Generation)

¾ Call Center: Option by Agent, Location Preference Distribution, and Vector Enhancements

¾ OPTIM/SCCAN

¾ Large Scale Meet-Me Conferencing

¾ System Capacity Wizard

¾ Support for ASG

¾ The ASA interface contains some very valuable applications that allow administrators to customize features and monitor system parameters:

¾ General Tab

¾ Administrators can manage users, stations, ports, and extensions.

¾ Advanced Tab

¾ Administrators can create templates, import and export data, and start emulations.




¾ Fault & Performance Tab

¾ Administrators can establish data polling schedules, and monitor trunks and alarms.

ASA General Tab

Avaya Communication Manager offers over 700 standard features, each offering enterprise-class functionality and usability. ASA provides comprehensive tools to manage users, stations, ports, and extensions; through ASA, the System Administrator is able to enable/disable features at the user level. ASA is not a web based tool

Since you are only offering the Standard Offering which only includes ASA, VAM, VMM and CNA you might not want to mention any of NMC or MSA or FPM.

Avaya MultiSite Administration (MSA)

An optional level above the functionality offered by ASA is Avaya MultiSite Administration (MSA) which is a browser based application that provides multi-user, graphical, web-based management of more than one Avaya media server and media gateway with extensive control over permissions of users and what they are allowed to configure. It provides centralized management of distributed networks and campus environments.

MSA Features

Avaya MultiSite Administration offers these powerful features:

¾ Enables multiple administrators to administer the same (or separate) Avaya media servers at the same time, remotely.

¾ Offers graphical station and system administration screens.

¾ Offers easy-to-use wizards for basic administration tasks.

¾ Offers detailed control of users and allowed functionality using Custom Privileges.

¾ Enables creation of templates that can be shared by all users across all systems

MSA Station Manager

¾ Station Wizard

◊ Creates new stations and the associated group memberships or makes basic changes to station. Creation of designation labels and addition of voice mail subscribers.

¾ Station Template

◊ Select the voice terminal type; select all features for the terminal. Make the template private to block general access or public so it can be used by any login.

¾ Move existing station by swapping with another station

¾ Like types of voice terminals can be swapped.




¾ Move multiple stations to new locations

◊ Extensions can be moved, new ports assigned, with other coverage points. If values for a field are left blank then current values for the extension will be preserved.

¾ Change voice terminal set types

◊ Change single or multiple stations to a new set type.

¾ Remove existing station(s)

◊ Remove single or multiple stations, extensions will be automatically removed from associated groups.

¾ Change station feature button(s)

◊ Change feature of button to be changed. If the button is left blank the value will remain as current in the switch.

¾ Move person among groups

◊ Administer membership of various groups. The wizard will remove extension(s) from groups and add to new designated groups.

Report Manager

¾ Analyze Configuration

◊ Carrier identifier

◊ Used slots

◊ Highest slot number used

◊ Number of slots

◊ Available slots

◊ Total carriers

◊ Total used slots

◊ Total available slots

¾ Print initialization options

◊ Correctly format printed reports or your printer.

¾ Look up extension

¾ Quickly search the database for an extension.

¾ Query station

¾ Search the database for information on extensions, search criteria can be:

◊ Name

◊ Extension

◊ Port

◊ Room




◊ Type

◊ Cor

◊ Cos

◊ Cable

◊ Jack

Fault and Performance Manager (FPM)

The optional Avaya Fault and Performance Manager (FPM) provides a network map or system view of the converged network and the tabular tools to monitor the status and performance of the devices on the enterprise’s network. The user can see into the network to examine faults and performance data from the Avaya Media Servers on the network.

FPM collects configuration, fault, and performance data from Avaya Proxy Agent or directly from an IP-enabled voice system using OSSI, and then displays the data in text, tables, and graphic formats.

The device data and user accounts are stored in the Avaya Integrated Management Database (IMD) and shared by the Avaya Fault and Performance Manager and the Avaya MultiSite Administration and Proxy Agent applications.

FPM operates on versions of Red Hat Enterprise Linux. It is available in the Avaya Integrated Management Release 3.0, System Management offer.




FPM Features

The primary features of the Fault and Performance Manager are:

¾ Graphical User Interface (GUI): The main window provides the following views of the managed nodes in the network, such as, system groups, DCS trunk connectivity, IP trunk connectivity, and clusters.

¾ Configuration: The users can view the configuration and administered properties of all supported systems (managed nodes) in both a graphic view and a table view.

¾ Administration: Users can define the system-wide parameters for the features, such as, data collection, exception logging, exception filtering, and exception alerting.

¾ Report Manager: Users can define the parameters for individual reports for all or selected systems. The report options include performance, configuration, and exceptions. The users can view the reports on screen in both the table and chart formats or direct the reports to a printer, HTML file, GIF file, or ASCII file.

¾ Scheduled Reports: The users can schedule reports to run on a daily, weekly, or monthly basis, and edit and delete schedules as needed.

Integrated Management Database

Integrated Management Database provides a single interface administration of users and configuration data of managed systems for MultiSite Administration, Fault and Performance Manager, and Proxy Agent. Integrated Management Database is an integrated management application with HTML pages viewable in Internet Windows 2000, XP and Windows 2003

Password Security Features:

¾ Authentication with customer AAA or LDAP server first withfailover to embedded login authentication.

¾ Force Admin Login Password Change (First use of default password for the “admin” login shall automatically prompt the user to change the password)

¾ Login password requirements (Passwords must be at least 8 characters in length; Length is administrable)

¾ Password Re-Use (Passwords cannot match any one of the previous four passwords)

¾ Password Contents (Passwords must contain at least one alphabetic character and one numeric character)

¾ Password Change Rate Limiter (Passwords cannot be changed more than once a day; Rate is administrable)

¾ Password Change Error Display (display error messages that provide guidance on why a password was not accepted)

¾ Password-Aging (passwords must be changed after 60 days; Cycle length is administrable)

for

Explorer 6.0 on




Communication Manager System Management

Proprietary protocol

Linux Server: MultiSite Administration Fault and Performance Manager Integrated Management Database SSG for adjunct and DEFINITY

Communication Manager

and SNMP

JAVA

Web Browser

alarming

© 2007 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. Under NDA 32

Avaya Voice Announcement Manager (VAM)

Avaya Voice Announcement Manager (VAM) is a stand-alone application. It is installed on a Microsoft Windows or Vista client PC and provides the ability to add, change, and remove recorded announcements and transfer them to Avaya media servers. Using WAV files, an announcement can be stored and sent over a LAN without file conversion. The WAV file repository simplifies storage and management of WAV files.

Users can view the current status of announcements at any time. Announcement files can be copied, backed up, or restored from Avaya media servers and gateways via the LAN. They can also schedule broadcasts and backups of announcements to single or multiple VAL boards and/or Media Servers.

Avaya VoIP Monitoring Manager

VoIP Monitoring Manager is a QoS monitoring and feedback software tool that supports realtime visualization of VoIP network performance from a client GUI application residing on the LAN or via remote access. VoIP Monitoring Manager receives Real-Time Protocol Control Protocol (RTCP) packets from IP phones and VoIP engines on media gateways. This allows the network manager to monitor a variety of call paths in real-time, including:

¾ Endpoint-to-Endpoint (end-to-end through the network)

¾ Endpoint-to-Gateway

¾ Gateway-to-Gateway (IP trunk monitoring over the WAN).

VoIP Monitoring Manager provides information on all QoS parameters using an easy-tounderstand graphical meter display. To support quick identification and localization of network issues that impact voice quality, VoIP Monitoring Manager can also generate traps based on preconfigured thresholds to any Network Management System.




The standards-based architecture of Avaya Integrated Management tools comply with accepted standards such as Simple Network Management Protocol (SNMP). This open architecture not only encompasses a wide range of Avaya communication devices and server, but also extends to integration with HP OpenView and products from Extreme Networks via integration with the Extreme Networks EPICenter network management application.

The VoIP Monitoring application is a diagnostic tool that monitors the quality of voice transmissions over the network. This product tracks transmissions of Real Time Control Protocol (RTCP) packets. The product listens to RTCP packets and populates the database with historical data.

You can query the database from a web browser to generate reports and to view real-time graphics of the RTCP packet transmission. The reports show active phones and other IP endpoints and identify the IP endpoints that are experiencing problems with jitter, delay, and packet loss.

The Avaya VoIP Monitoring Manager provides the ability to monitor VoIP network quality, by receiving QoS statistics from Avaya IP end-points and providing the means to visually display the real-time data.

Beyond trouble-shooting and monitoring, VoIP Monitoring Manager is also a pro-active notification solution that warns the network manager of potential degradation of voice quality in their network – providing valuable time to identify, localize, and fix issues in the

network as they arise. In this scenario, VoIP Monitoring Manager can be configured to automatically send SNMP (Simple Network Management Protocol) traps to a Network Management System (NMS) based on a number of QoS threshold policies.

Avaya Fault and Performance Manager provides a hierarchical view of the health and status of all voice systems in the network, with drill-down capabilities to display information on configuration, fault, and health data of individual systems and adjuncts in text, tabular, and graphical formats.

Fault and Performance Manager is also a powerful reporting tool, with system-wide administration of parameters that allow the network manager to quickly define settings for data collection, logging, and alert levels, and a flexible report manager that provides detailed information on performance, configuration, and

exceptions/alarms. To help simplify interpretation of system messages and alarms as they appear, an integrated helpdesk feature automatically launches the Communication Manager maintenance manual with the specific alarm in focus. The ability to set up filtering of alarms on a system wide or individual system basis makes finding alarms and elevating the alert status a valuable trouble shooting tool. Alerts can be easily emailed or forwarded on to any SNMP compliant application




Fault and Performance Manager

Fault and Performance Manager Help Desk Functionality

The Maintenance Manual is By clicking on Helper displayed, providing detailed

Assistant information on that specific alarm

37© 2007 Avaya Inc. All rights reserved. Avaya – Proprietary & Confidential. Under NDA

Avaya Network Management Console

Avaya optional Network Management Console provides the customer with a product operates standalone as the customer’s main management console. The Network Management Console provides discovery of IP devices on the customer’s network, fault monitoring of those voice systems, a focal point for viewing and launching element managers, as well as network wide provisioning and configuration applications that enable the customer to manage their network of devices. These applications include the Avaya Network Management Console with VOIP view, Avaya Network Configuration, Avaya Software Update Manager and Provisioning and Installation Manager. The Avaya Network Management Console supports converged network environments composed of multi-vendor equipment (This encompasses only major devices from key vendors such as Avaya partners). The Avaya Network Management Console and applications supports all Avaya IP based voice and data devices, wired and wireless, to create a full convergence solution.




The Avaya Network Management Console shown above provides discovery of IP devices including voice elements such as Avaya S8300 Server, Avaya S8730 Server, TN2501AP boards and IP phones placing the discovered devices on a network map, as illustrated below.

Note: Although optional with Communication Manager standard software Avaya Network Management Console is an Entitlement for the Enterprise version.

6.1.0 Administration Functions The proposed systems management solution must support: station user moves, adds, and changes; trunk group definitions and individual trunk circuit programming; voice terminal parameters; call restriction assignments; class of service definitions and assignments; password resets; customer profile database; ARS routing tables; group definitions and assignments; first digit tables; dial plan; feature access codes; paging/code call zone assignments.


Confirm the proposed systems management solution supports each of the listed administrative functions. Identify any functions not supported.




Avaya Response:

All above functions and more are supported. Avaya Integrated Management will allow VoiceCon to administer business site moves, adds and changes; configure distributed networks and campus environments; and manage performance in converged networks. Administrators can use the LAN to transfer recorded announcements to Avaya Servers and Gateways, and managing a network of multiple spare processors over ATM wide area networks - it enhances system reliability and business continuity.

The Avaya Station Administration Wizard is accessible from the Standard Management Solutions home page on the media server. With this easy-to-use wizard, VoiceCon can quickly add, move, or change data on a station.

Group Assignments The administration subsystem must support each of the following group definitions and assignments

• Abbreviated Dialing (System, Group, Enhanced)

• Hunt Groups

• Call Coverage Answer Groups

• Pickup Groups

• Intercom Groups

• Terminating Extension Groups

• Trunk Groups


Confirm administration support for each of the listed group definitions. List any and all groups not supported by the administration subsystem.

Avaya Response:

Comply. All of the group definitions listed above are supported by the system administration applications. Descriptions of the features are provided in Appendix 1, Avaya Communication Manager Feature Descriptions.

6.1.2 Facilities Performance Management & Reports The management system must be able to collect, analyze, and provide reports for a variety of system operations.

Avaya Response:

Read and understood




6.2.1 Basic Trunk Usage and Traffic Trunk traffic records should be kept for all inbound and outbound calls, identifying the trunk group and trunk channel, time and duration of call.


Confirm that the proposed facilities management system satisfies this requirement.

Avaya Response:

Comply. The standard reporting capabilities of Avaya Communication Manager include extensive traffic reporting capabilities on trunk groups. The reports are included in the base application, and do not require additional servers or software.

There are several reports available for trunk groups, including:

¾ The Trunk Group Summary report gives traffic measurements for all trunk groups except for personal central office line groups. By using this report, you can determine the trunk group total usage (in CCS), the total number of calls, trunk blockage, and other measurement data.

¾ The Trunk Group Hourly report provides data necessary to validate the information in the Trunk Group Summary report and to size the trunk groups. A separate report is generated for each trunk group.

¾ The Trunk Out of Service report lists up to a maximum of five trunks (in each trunk group) out of service when sampled. The number of times the trunks are out of service when sampled is also given. The trunk outage data is kept for the current day, the previous day, and the last hour.

¾ The Trunk Group Status report gives a current indication of the load on various trunk groups in terms of the number of calls waiting to be serviced. For each trunk group, the Trunk Group Status Report displays the number of calls in the queue waiting to be serviced. For comparative analysis, the trunk members in the group active on calls are also displayed. With this data, it is possible to rearrange the members in the groups to provide load balancing. For example, if one group shows a higher number of calls waiting in the queue and the size of the group is too small; more members can be added to that group.

¾ The Call-by-call (CBC) Trunk Group Measurements report displays last-hour traffic data for any specified Call-by Call trunk group, provided the trunk group had a Usage Allocation Plan (UAP) administered for the last-hour. Use the report to monitor the trunk group and to determine if the UAP meets current needs. Whenever it is determined changes are required, you must make these changes on the appropriate trunk group screen(s).

¾ The Trunk Lightly Used report lists the five trunk members with the lowest number of calls carried for each trunk group. The trunk lightly used data is kept for the current day, the previous day, and the last hour.




6.2.1.1 Individual Trunk Line Counters Vendor Response Requirement

Confirm that individual trunk line counters measure and report: Number of call attempts; Number of blocked trunk lines; Traffic intensity (Erlangs).

Avaya Response:

Comply. The Trunk Group Summary report gives traffic measurements for all trunk groups except for personal central office line groups. By using this report, you can determine the trunk group total usage (in CCS), the total number of calls, trunk blockage, and other measurement data.

6.2.1.2 Outgoing Trunk Route Counters Vendor Response Requirement

Confirm that outgoing trunk route counters measure and report: Number of outgoing attempts; Number of successful calls overflowing to another route; Number of lost calls due to blocking; Number of blocked trunks in measurement; Traffic intensity (Erlangs).

Avaya Response:

Comply. Several outgoing trunk route measurements are available across various reports; please refer to the screen shot provided in item 6.2.1.1.

6.2.1.3 Incoming Trunk Route Counters Vendor Response Requirement

Confirm that incoming trunk route counters measure and report: Number of incoming call attempts; Number of trunks in the measurement; Number of blocked trunks in the measurement; Traffic intensity (Erlangs).

Avaya Response:

Comply. Several outgoing trunk route measurements are available across various reports; please refer to the screen shot provided in item 6.2.1.1.




6.2.1.4 Both Way Trunk Route Counters Vendor Response Requirement

Confirm that both way trunk route counters measure and report: Number of incoming call attempts; Number of trunks in the measurement; Number of blocked trunks in the measurement; Traffic intensity (Erlangs).

Avaya Response:

Comply. Please refer to the screen shot provided in item 6.2.1.1.

6.2.2 Attendant Consoles Attendant counters should measure all attendants in the system, or individual attendant positions. Record measurements include: number of answered calls; number of calls initiated by attendant; accumulated handling time for all calls; accumulated handling time for recalls; accumulated handling time for calls initiated by attendant; accumulated total delay time for recalls; number of answered recalls; number of abandoned attendant recalls; accumulated waiting time for abandoned calls to an attendant; accumulated waiting time for abandoned recalls, and accumulated response time for all types of calls.


Confirm that attendant counters measure and provide reports for each of the listed parameters. Identify attendant parameters which are not measured.

Avaya Response:

Comply. The proposed solution includes inherent detailed attendant reporting. No additional hardware or software is required to support the feature. The attendant group reports are used to assess the quality of service provided customers calling through the listed directory numbers and to facilitate the management of the attendant group so it is neither under – nor over-staffed. Attendant group measurements appear on two reports.

¾ The Attendant Group report provides hourly traffic measurements for the attendant group as a whole.

¾ The Attendant Positions report gives peak individual attendant position measurements.

Both reports are available as PEAK reports for yesterday’s peak hour, today’s peak hour, and the last hour. A peak hour is the hour within a 24-hour period with the greatest usage (Time Talk plus Time Held) for the specified day. Hourly data for the entire attendant group can be obtained by polling the Attendant Group Report on an hourly basis.

Attendant Group Measurements Report




Attendant Positions Measurements report

The Attendant Positions Measurements report provides hourly individual attendant position measurements. It is used to assess personnel performance, and to identify when additional training may be necessary.

Attendant Positions Measurements Report – page 1

Attendant Positions Measurements Report – page 2

Attendant Speed of Answer report

The Attendant Speed of Answer report gives the console attendant group average speed of answer for each hour of a 24-hour period, for either yesterday or today.




Attendant Speed of Answer report — page 1

Attendant Speed of Answer report — page 2

6.2.3 Stations Station counters should measure individual stations or station group traffic statistics, including: number calls; number of stations in measurement; number of blocked stations in measurement; traffic rating (Erlangs).


Confirm that station counters measure and provide reports for each of the listed parameters. Identify station parameters which are not measured.




Avaya Response:

Comply. Avaya VoIP Monitoring Manager, in combination with the reports resident on the server, provide all of the functions listed above.

6.2.4 Traffic distribution When applicable, traffic distribution across the internal switching network should be measured for each local TDM bus, traffic over each highway bus, and traffic across the center stage switch by each switch network interface link.


Confirm that traffic distribution is measured and reported for each switch network element listed. Identify what is not measured and reported.

Avaya Response:

Comply. Traffic distribution is measured by the VoIP Monitoring application described in responses for 6.03 and 6.05 of this proposal.

6.2.5 Busy hour traffic analysis Busy hour traffic analysis measurements for trunks, stations, and the internal switch network should be performed and reported for any one hour interval for any time of the day.


Confirm busy hour traffic measurements for trunks, stations, and the internal switch network for any one hour interval for any time of the day.

Avaya Response:

Comply. Please refer to the busy hour reporting capabilities discussed in responses for 6.2.1, 6.2.1.1 and 6.2.2.

6.2.6 Erlang Ratings Erlang rating should be calculated and reported for individual trunk lines, each trunk group, and all trunk groups. CCS ratings should be calculated for individual stations or groups of stations.


Confirm Erlang and CCS rating calculations and reporting for each listed item.

Avaya Response:

Comply. Erlang ratings are provided on the trunk group reports discussed in responses for section 6.




6.2.7 Processor Occupancy System call processing performance is measured in terms of Busy Hour Calls (Attempts and Completions). The percent of maximum call processing capacity should be reported for programmed time intervals. Threshold reports should also be generated to monitor system load factors.


Confirm measurement and reporting of processor occupancy and threshold levels

Avaya Response:

Comply. The proposed Avaya solution supports several processor occupancy reports, defined as the percentage of time the configuration’s processor is busy performing call processing tasks, maintenance tasks, administration tasks, and operating system tasks. As a contrast, the percentage of time the processor is not used is referred to as idle occupancy.

The primary objectives of the processor occupancy reports are:

¾ To provide a summary of customer usage data so processor occupancy and available capacity can be determined.

¾ To display, on a per time interval basis, the processor occupancy and associated calling rates which facilitates the isolation of certain customer reported problems.

There are four different processor occupancy commands:

¾ list measurements occupancy summary

¾ list measurements occupancy last-hour

¾ list measurements occupancy busiest-intervals

¾ list measurements communications-links

The first three commands provide processor occupancy data and associated call traffic for different measurement intervals. The last command provides a picture of the traffic data generated on each processor interface link.

6.2.8 Threshold Alarms For a variety of system hardware devices it should be possible to define a congestion threshold value, and measure generated alarms. Alarms are recorded in an Alarm Record Log. The types of devices that can be tracked include: tone receivers; DTMF senders and receivers; conference bridges; trunk routes; modem groups.


Confirm recording and reporting of alarms for each listed item.

Avaya Response:

Comply. Alarms are logged in the Hardware Error Report and can be reviewed on a regular basis. The administrator can view all active system errors on the error log. The user can also specify a particular component of the system or a certain time period to be reported on the error log.




This example shows errors on trunk routes and digital line cards; however, all error types are tracked.

6.2.9 Feature Usage Feature usage counters for selected station features, e.g., call forward, call transfer, add-on conference, and attendant system features, e.g., recall, break-in, should be measured and reported for programmed intervals.


Confirm recording and reporting of feature usage counters for both station and attendant operations.

Avaya Response:

Comply. There are several reports that provide measurement information about features. Including the following:

The following reports provide information about the Announcements feature:

¾ Denial Events Log – if a working VAL announcement file is deleted over FTP, the next attempt to play the announcement fails, and the system adds a software event to the Denial Events Log. Users can view the Denial Events log to see if the announcement was deleted, and to see if other events occurred that are related to announcements.

¾ Announcement Measurements – users can view a report of announcement measurements. This report includes how many times an announcement was queued to play, how many callers dropped while in the queue, and how many times all announcement ports were busy during the report period.

The ACA Measurements Report (ACA) shows the audit trail for ACA calls.

The following reports provide information about the Call Coverage feature:

¾ The Coverage Path Measurement report shows coverage activity about the coverage paths.

◊ The Principal Coverage Measurement report shows coverage activity about the called extensions.

◊ The Call Detail Recording (CDR) report shows the outgoing trunk calls.




The following reports provide information about the Enhanced 911 feature:

¾ The Emergency Access Calls report shows the following information for each emergency call:

◊ Extension

◊ Event

◊ Type of call

◊ Time

The following reports provide information about the Security Violation Notification feature:

¾ The Security Violations Status Report shows details of the last sixteen violations of each type. The Barrier Code and Authorization Code reports also include the calling party number from which the attempt was made, if that information is available.

The following reports provide information about the Station Hunting feature:

¾ The List Usage report shows all the extensions that use an extension as the huntto-station.

¾ The Uniform Dial Plan report shows the details of the Dial Plan.

6.2.10 VoIP Monitoring The management system should collect and store data to track usage and performance data of IP gateway devices, IP phones, and VoIP intercom/trunk calls. VoIP information reports may include: tracking of IP gateway devices and calls that pass through each gateway; gateway congestion; assignment of services or routes to gateways; tracking of phone numbers dialed or originating off-site numbers; and IP gateway addresses.


Briefly describe all VoIP monitoring information records and reports that are available. Specify if VoIP QoS parameters such as jitter, call delay/latency, and packet loss are tracked and reported, and if a system administrator can monitor VoIP calls in real-time for QoS observing? Indicate if any third party equipment is being proposed as part of your solution.

Avaya Response:

Avaya VoIP Monitoring provides the requested features, as described responses for 6.03 and 6.04 in this proposal. In addition, the proposed solution includes six reports that list the activity on your IP media processor circuit packs, useful for managing both intra-switch traffic, as well as IP trunking. These reports are described below:

¾ IP Codec Resource Hourly report - lists the codec resources used on all IP media processors for the last 24 hours, from the current hour backwards, for a specific region. This report lists separate information for the G.711 Codec’s and the G.723/G.729 Codec’s.




IP Codec Resource Hourly report

¾ IP Codec Resource Summary report - lists the codec resources used on all IP media processors for a specific peak hour for all regions. You can list reports for yesterday’s peak, today’s peak, or the last hour. This report lists separate information for the G.711 codec’s and the G.723/G.729 codec’s.

IP Codec Resource Summary Report

¾ IP Codec Resource Detail report - lists the codec resources used on all IP media processors for a specific peak hour for a specific region. You can list reports for yesterday’s peak, today’s peak, or the last hour. This report lists separate information for the G.711 codec’s and the G.723/G.729 codec’s.

IP Codec Resource Detail report




¾ IP DSP Resource Hourly report - lists the codec resources used on all IP media processors for the last 24 hours, from the current hour backwards, for a specific region.

IP DSP Resource Hourly report

¾ IP DSP Resource Summary report - lists the codec resources used on all IP media processors for a specific peak hour for all regions. You can list reports for yesterday’s peak, today’s peak, or the last hour.

IP DSP Resource Summary Report

¾ IP DSP Resource Detail report – lists the codec resources used on all IP media processors for a specific peak hour for a specific region. You can list reports for yesterday’s peak, today’s peak, or the last hour.




6.3 Optional Reports Directory records may include each subscriber’s name along with a variety of phone numbers such as primary, published, listed, emergency, and alternate, as well as authorization code information, job title, employee number, current employment status and SSN.

Inventory records and management is used to administer any kind of inventory product part, including: PBX common equipment (cabinets, carriers, circuit cards); voice terminals and module options; jacks, and button maps. The reports allow administrators to accurately re-charge items. Inventory can be tracked by data such as user, system (PBX or other networks), jack, serial number, asset tags, trouble calls, recurring and non-recurring costs, and general ledger codes. The inventory management system may also include records containing the following data: purchase date, purchase order number, depreciation, lease dates, manufacturer and warranty information.

Cabling records keep track of all cable, wire pairs, distribution frames, wiring closets and all connections (including circuits) down to both the position and the pair level. Cable records include starting and ending locations, description, type and function. Individual cable lengths are maintained and automatically added, as is the decibel loss, for the entire path. Information can also be provided on the status of all cable runs, as well as the number of pairs it contains, the status of the pairs, and the type of service it provides.


Identify and briefly describe your proposed management system’s Directory, Inventory, and Cabling reports, if available.

Avaya Response:

The optional reports can be provided by one of our Dev-Connect Partners that can integrate LDAP and inventory control with cable records, pricing for these options have not been provided in the Avaya Solution

6.4.0 Call Detail Recording Call Detail Record (CDR) data should be compiled for all successful incoming and outgoing trunk calls. Call record fields typically include the following:

• Date

• Time

• Call Duration

• Condition Code (categorizes information represented in the call record)

• Trunk Access Codes

• Dialed Number

• Calling Number




• Account Code

• Authorization Code

• Facility Restriction Level for Private Network Calls

• Transit Network Selection Code (ISDN access code to route calls to a specific inter-exchange carrier)

• ISDN Bearer Capability Class

• Call Bandwidth

• Operator System Access (ISDN access code to route calls to a specific network operator)

• Time in Queue

• Incoming Trunk ID

• Incoming Ring Interval Duration

• Outgoing Trunk ID


VoiceCon will purchase its own third party call accounting and billing system. Identify all available CDR reports that can be generated for any or the entire call record field data listed above.

Avaya Response:

Avaya Site Administration can generate call accounting data that is needed by call accounting and billing systems. The following categories can be placed into an export file that can be imported into a third party billing system:

¾ agent login-id,

¾ authorization codes,

¾ station data,

¾ trunk-groups

¾ circuits

The Call Detail Recording feature of Communication Manager records detailed call information on incoming and outgoing calls for the purpose of call accounting, and sends this call information to a Call Detail Recording (CDR) output device. You can specify the trunk groups and extensions for which you want records to be kept as well as the type of information to be recorded. You can keep track of both internal and external calls. This application contains a wide variety of administrable options and capabilities.

Two types of formats are sent to the CDR output device, date record and call detail formats. You can use the customized record format to make up your own call record. You can determine the data elements you want and their positions in the record. This method may be necessary if you want to include certain data elements that are not available on the standard formats. However, whatever device you use to interpret the CDR data needs to be programmed to accept these formats. Consult your Avaya representative before using a custom record format.




The following data elements are available:

Data Field Description

Time of day-hours Authorization code Dialed number

Time of day-minutes Feature flag Calling number

Duration-hours Access code used Account code

Duration-minutes Outgoing circuit ID FRL

Duration-tenths of minutes Access code dialed IXC

Condition code Incoming circuit ID

6.5.0 Maintenance System maintenance operations should, at minimum, support the following:

• Monitoring of processor status

• Monitoring and testing of all port and service circuit packs;

• Monitoring and control of power units, fans, and environmental sensors;

• Monitoring of peripherals (voice terminals and trunk circuits);

• Initiate emergency transfer and control to backup systems;

• Originate alarm information and activate alarms.


Confirm support for the required maintenance monitoring activities by completing following table:

Activity Yes/No

Monitoring of processor status Yes

Monitoring and testing of all port and service circuit packs Yes

Monitoring and control of power units, fans, and environmental Yes sensors

Monitoring of peripherals (voice terminals and trunk circuits) Yes

Initiate emergency transfer and control to backup systems Yes

Originate alarm information and activate alarms Yes

Avaya Response:

Comply. All maintenance and monitoring activities listed above are supported. It is important to understand that internal systems diagnostics play an important role in the reliability of a system. Avaya Communication Manager has more than 30 percent of its lines of code dedicated to internal diagnostics to maintain constant vigilance over the health and well being of the system. It is these diagnostics that fundamentally give the Avaya solution its advantage over many other systems because in conjunction with these diagnostics comes a patented system of artificial intelligence we call Avaya EXPERT Systems Diagnostic Tools.


MINOR, or



6.5.1 Alarm Conditions There are usually several types of communications system alarm conditions: Major, Minor, and Warning.


Briefly describe how your management system defines a Major, Minor, and Warning alarm.

Avaya Response:

The Communication Manager application monitors and logs alarms. Alarms are classified as major, minor, or warning, depending onthe degree of severity and the effect on the system.

The system keeps a record of every alarm that it detects. This record, the alarm log, and the error log can be displayed locally on the management terminal. An alarm is classified as MAJOR, WARNING, depending on its effect on system operation. Alarms are alsoclassified as ON-BOARD or OFF-BOARD.

¾ MAJOR alarms identify failures that cause critical degradation ofservice and require immediate attention. Major alarms can occur on standby components without affecting service, since their active counterparts continue to function.

¾ MINOR alarms identify failures that cause some service degradation but do not render a crucial portion of the system inoperable. The condition requires attention, but typically a minor alarm affects only a few trunks or stations or a single feature.

¾ WARNING alarms identify failures that cause no significant degradation of service or failures of equipment external to the system. These are not reported to the Avaya alarm receiving system or the attendant console.

Alarms are further classified as:

¾ on-board problems originate within the circuitry of the alarmed circuit pack

¾ off-board problems originate in a process or component that is external to the circuit pack

Feature button lamps on any phone can be administered to act as alarm indicators, similar to the alarm lamp on the attendant console.

6.5.2 Maintenance Reports Vendor Response Requirement

Provide a list all standard maintenance alarm reports provided by your management system.

Avaya Response:

If a maintenance object in the system begins to fail some of the periodic tests, the system automatically generates an alarm that indicates the system needs to be restored to a normal condition.

Alarms are communicated to the system users and technicians by entries in the alarm log and the lighting of LEDs located on the attendant console, on all circuit packs, on the server, on the Cajun Ethernet switch (if used), and, optionally, on telephones designated by VoiceCon’s System Administrator. Warning alarms are not reported to the attendant console.




The system alarm log is available for viewing via the administrative terminal. In addition, Avaya Site Administration provides an alarm monitoring capability, as illustrated below:

6.5.3 Remote Maintenance Vendor Response Requirement

Briefly describe the available options used to support remote maintenance operations for both customer access and for an outside maintenance service provider. Specify how the system alerts a remote service center when an alarm condition occurs, the trunk circuit requirements for alert transmissions, and security measures to prevent unauthorized access.

Avaya Response:

Avaya can maintain systems from Global Services Delivery centers by means of a connection to the remote maintenance port on your system. The local interface and the remote terminal provide access to maintenance and administration procedures that allow Avaya to diagnose your system, isolate faults, and repair the system. In order to provide remote diagnostics and resolution to your applications, Avaya requires connectivity to your communications systems. This can be achieved through the use of Secure Access and Control, which is a device that can monitor up to 500 server/devices on your network through a secure VPN tunnel, or you may elect to install a dedicated phone line, INitialization and ADministration System (INADS) line, to each applicable server. The INADS line or Secure Access and Control server enables Avaya to troubleshoot and assess your application and provide help desk assistance.




When an Avaya Communication Manager enabled system recognizes not just outright failures, but any system anomaly that might be service affecting, the application performs a “cry for help” by out calling to Avaya Global Services Delivery (GSD) center. When the call is received a trouble ticket is created while the alarm in question is automatically routed to EXPERT Systems Diagnostic Tools to begin trouble resolution. With millions of hours of actual inservice information from the largest, longest installed base of telephony platforms in North America, EXPERT Systems patented algorithms are able to make decisions on how to resolve the alarm condition and in combination with the GSD remote engineers, clears over 98% of all system-generated alarms remotely.

To assist in rapid problem diagnosis and resolution, Avaya requires that customers provide the connectivity for remote maintenance administration access for each applicable and supportable system.

Avaya Service Agreement customers have added protection against unauthorized access and toll fraud. Avaya provides unmatched service entitlements, designed to proactively protect your system, for example:

¾ Fraud intervention - if you suspect fraud is occurring real-time, call the PBX Security Service Team on the Hotline at 1.800.643.2353. An Avaya representative will log into your system and assist you in shutting down fraud. Reducing the duration of fraud saves you money. Incident Response is available 24 hours a day, 365 days a year.

¾ Quarterly Advisory Letter - Allow businesses to take a proactive stance on security measures by scanning selected areas of the system each quarter.

¾ Security advisories and handbook – available on the customer support web site at http://support.avaya.com

Additionally, Avaya has Communications System Security options available.

6.6.0 Provisioning All services should be provisioned in one step. Services should include station configuration, voice mailbox configuration, E-911 location, billing attributes, directory attributes, and mobile Email attributes (Blackberry) and the configuration of other end user applications.

For example, if your solution includes a zone paging application, the ability to assign a station to a zone and change the zone membership as a whole must be accessible through the configuration (provisioning) interface.

Templates must be supported to organize different settings across different systems according to organizational need. At a minimum, the voice station configuration and the associated voice mailbox must be provisioned in one step through one interface.

Your proposed provisioning application or interface must create a complete audit trail and must allow groups of changes to be scheduled for a future time. Further, the solution must support mass create, delete and modify functions to support bulk operations.


Describe the provisioning workflow you recommend showing how each of your proposed solution components is utilized. List any functions above which are not available. List any systems or devices which are not now part of your provisioning interface and provide a roadmap statement of how you will treat this situation going forward.




Avaya Response:

A fully integrated single source-provisioning tool is currently not available. However, Avaya’s various administration tools offer many time and cost saving benefits. These tools include the Avaya site Administration (ASA), Multi-Site Administration (MSA), the Avaya Provisioning and Installation Manager (PIM), and third party applications using Avaya’s Directory enable Management application. Each provides specific capabilities that will benefit any enterprise.

The ASA and MSA tools provided template and wizard capabilities to allow for easy addition of users. This includes the addition of voice mailboxes and would incorporate pre-designed images to help expedite the installation process.

Avaya Provisioning and Installation Manager (PIM)

PIM can help an enterprises save time and money on large-scale branch office deployments by replacing error-prone and time-consuming device-by-device provisioning with a tool that enables the definition of customized templates that can be defined centrally and applied to large groups of gateways at one time. Designed to integrate as part of the staging and installation process, Provisioning and Installation manager allows network experts at the staging center to quickly gather information using Electronic Pre-Installation Worksheets that can then be imported into the application to create configuration templates and individual device profiles to record unique parameters for each gateway. These templates and profiles can be stored in advance as part of the initial staging process, and then applied to groups of gateways as part of the bulk provisioning process during actual installation. The result is large-scale branch office deployments performed faster, with less error, and with reduced need for technical expertise at the installation site. Provisioning and Installation Manager can also be used for ongoing administrative tasks that require configuration changes to groups of gateways.

Pre-Staging and Bulk Provisioning of Branch Office Media Gateways

Avaya Provisioning and Installation Manager (PIM) supports:

¾ Template-based bulk provisioning

¾ Electronic Pre-Installation Worksheets with direct import into the application to create configuration templates and device profiles

¾ Scheduled provisioning of gateways at pre-stage or during actual installation

¾ Time-consuming device provisioning is replaced with a bulk provisioning process that is faster and less expensive

¾ Electronic Pre-Installation Worksheets can be directly imported into PIM, enabling deployments to be performed faster and with error

Advanced provisioning reduces the need for technical expertise at the installation sites.

Third Party Applications

There are a variety of Avaya DevConnect partners who have developed and can customize more enhanced “all-in-one” tools using Avaya’s Directory Enabled Managements application that provide LDAP level support.

Avaya continue to evolve the Integrated Management Tool Suite to incorporate greater and greater functionality and strives to reach an “all in one” solution in the future. No specific roadmap is available for distribution.



Part 1 - Section 7 - Integrated Messaging Systems

7.0.0 Integrated Messaging System VoiceCon requires a CHQ-based voice messaging system that must be fully integrated with the proposed IPTS network solution. VoiceCon also requires integration of the proposed voice messaging system with a Microsoft Exchange messaging system to provide “unified” messaging applications. The proposed voice messaging system solution must be centrally located at the VoiceCon CHQ location, and be capable of supporting station users at the twol remote VoiceCon facilities (RO and SO).

The voice mail system will also serve as an automated attendant position for select incoming trunk calls, and also as a secondary point of coverage as an automated attendant system for designated stations. All software and hardware necessary to interface with the existing telephone system will be provided under this bid.

The sizing requirements are: Installed/Equipped Capacity Maximum Capacity

Number of Subscribers 2000 3,000 Number of Ports 96 128 Hours of Storage 1000 1500

Automated attendant ports are included in the requirements. A Grade of Service level of P.01 is required.


Briefly describe the proposed integrated messaging solution, and provide details about the voice mail system architecture and it’s interconnection to the voice communications system and Microsoft Exchange system. Include processing system platform information in the discussion. Verify that the system being bid can comply with each of the proceeding requirements.

Avaya Response:

Comply. The Avaya Modular Messaging solution is a powerful IP- and standards-based voice messaging platform designed for single- or multi-site global enterprises, centralized or decentralized. It offers exceptional scalability and an extensive feature package of call answering and messaging capabilities. Messages are accessible any time, anywhere from a wide array of access devices including telephones, PC graphical user interfaces, and PDAs. Graphical user capabilities, explained in detail in response to 7.4.4, provide a means of unifying voice, fax, text and email messages to be viewable, created, and acted upon from within the MS Exchange Outlook client. Optionally, Avaya Modular Messaging for Microsoft Exchange is also available as a single mailbox solution whereby all voice, fax, text, and email messages are stored within a single inbox.

Using the proposed Avaya one-X Speech application, mobile professionals can stay connected to customers, associates and partners via speech commands, wireless devices, and web browser. They can get to their most important information and communications functions through their phones, PDAs, desktop PC, or laptop—choosing the device that is most suitable at that time.

We are proposing the Avaya Modular Messaging with an Avaya Message Storage Server. This configuration contains Avaya Messaging Application Server (MAS) units with a Windows operating environment and an Avaya Message Storage Server (MSS) with a Linux operating environment. For increased security,

performance, control, and application management, the MSS is connected to the Avaya Messaging Application Server through a private Ethernet local area network (LAN). The MSS is designed so that it has no administratively activated offline mode. One-X Speech consists of an industry-standard server with Avaya one-X Speech Access software.




All new installations of the Avaya Messaging Application Server software reside on the Avaya-provided S3500 server platform running Microsoft Windows 2003 Server Application Kit with Service Pack 2. All new installations of the Avaya Modular Messaging Message Storage Server software reside on the Avaya S3500 platform with Red Hat Enterprise Linux Version 4.0 as the operating system.

With T1 QSIG integration, one digital pathway between the Avaya™ PBX and the Avaya Message Application Server (MAS) transmits both call information and voice communications. The pathway is provided by an ISDN digital link (QSIG), which provides channels that connect to the Dialogic T1 card. Within the D-Channel, routing information is sent to the MAS containing information regarding the source of the call with reason codes. The MAS processes call information from the supplementary code in the D-Channel, which routes call reasons directly to mailboxes.

Message-Waiting indication is set and canceled using the supplementary code service. Voice is carried through the system in digital format.

Regarding capacities, the IP connection to the optional Avaya S8500C supports up to 144 ports and 20,000 users. Up to 15,000 GSM hours/3,000 G.711 hours is available on the MSS. Up to 5 Avaya Modular Messaging Application Servers are supported in a voice mail domain. A voice mail domain can serve a network of switches, provided the Administrator ensures that the network uses a single switch as a gateway to the voice mail domain. Using the voice mail domain construct, up to 500 networked nodes can support up to 250,000 subscribers.

An Automated Attendant is a fully integrated Modular Messaging feature used for routing callers who have reached the organization’s access number to their desired extension number. The Automated Attendant interface greets callers and guides them through the process of entering the extension number of the called subscriber. Administrators can configure Automated Attendant to allow callers to identify the called subscriber by spelling the subscriber’s name using touchtone keys. The Automated Attendant interface also allows callers to reach designated system operators provided that an active schedule exists for the covering extension.

Caller Applications are a collection of menus and prompts that allow administrators to extend those parts of the Modular Messaging TUI that are accessible to callers. Using Caller Applications, administrators can extend the system Automated Attendant and the Common Caller Interface functionality, depending on the requirements of the organization. A Caller Application is not a mailbox and does not require that a mailbox be created on the message store or be dedicated to its use. A Caller Application contains one or more nodes, each of which can interact with the caller and pass control to another node. Up to 120 Caller Applications are supported per voice mail domain.


integrations



Caller Applications are separate applications—such as complex Automated Attendants, listen-only mailboxes, and bulletin boards—which can be designed using a Microsoft Windows graphical user interface (GUI)-based editor tool that is deployed across voice mail domains. Caller Applications can be used to accomplish most of the same functions as Automated Attendants (including nested Automated Attendants).

Some basic functions that a Caller Application can provide include:

¾ Transferring callers to a specified mailbox

¾ Allowing callers to record messages

¾ Sending messages to either a mailbox number or an e-mail address that is configured as part of the Caller Application

¾ Providing directory assistance for callers to use the dial-by-name functionality of the TUI

7.1.0 Support for Open Standards Vendor Response Requirement

• Describe voice messaging system’s support for open standards.

• List the clients that can be used with your proposed solution.

• For proprietary clients, detail minimum hardware and software requirements

Avaya Response:

Avaya Modular Messaging supports the following industry open standards:

¾ Intel processors

¾ Dialogic Tip/Ring boards, Dialogic T1 and E1 port boards, and Dialogic Digital Set Emulation (DSE) port boards. IP integration does not require port cards. All are possible integration methods.

¾ Linux operating system (in Avaya MSS configurations) and Microsoft Windows operating system (Avaya MAS)

IP and Internet Standards – These include IP for server-to-server transport, IP Networking, IMAP4 and POP3 client access to messages (available in other configurations but not required for the proposed solution), SMTP/MIME for sending and receiving messages, and LDAP for attribute storage (user and system data) and directory queries (name, number, address).

Switch Integrations – These include H.323-based IP integration, Q.Signaling (QSIG), Enhanced Inband Analog, RS232 for serial switch (SMSI/SMDI), and Digital Set Emulation (DSE).

Because Avaya has developed Modular Messaging as a standards-based platform, significant flexibility is introduced with regards to switch integrations developed and supported. This supported mix contains IP-based integrations, digital technology (including T1/E1 and digital set emulation), and a variety of analog-based technologies. Additionally, there are numerous integrations written, but not yet generally available based on customer demand.




Future development will also be based on customer need and demand, as well as technological advances in this area of development. At any given time, any combination of these integrations can be supported within a customer's Modular Messaging network, combining many of the developed integrations to support a broad mix of switching environments throughout the country and world.

Audio Encoding Formats – These include Global System for Mobile Communications (GSM) and G.711 (A-law and µ-law).

Other Standards – These include standards established by the government and standards bodies for compliance areas. These include Product Safety, Electro Magnetic Compliance (EMC), and Telecommunications.

Supported Clients

Avaya Modular Messaging includes IMAP4 access to messages from user client software packages. Avaya has conducted successful interoperability testing with Microsoft Outlook 2007, Microsoft Outlook 2002, Microsoft Outlook 2000, Microsoft Outlook Express 5.0, IBM Lotus Notes R6, and IBM Lotus Notes R5.

7.1.1 Security Features Vendor Response Requirement

Describe all security features available with the voice messaging system to prevent abuse and unauthorized access.

Avaya Response:

Comply. Avaya is acutely aware of the need for stringent security measures to be in place to minimize vulnerability. These areas include user mailboxes, system administration procedures, and limiting caller transfer capabilities.

Mailbox Security

The Administrator initially configures user’s mailboxes with a default password or a different randomly generated password for each new mailbox. On the first login from the telephone user interface, the user can be required to set a new numeric password between the Administrator-defined minimum and 15-digit maximum length. The System Administrator establishes the minimum password length as a system wide parameter. The password cannot match the mailbox number, include leading zeros, match up to the previous 20 passwords, contain consecutive digits, or use all the same digits. As an extra security measure, the Administrator can choose to force users to press [#] after entering their mailbox passwords. Access to messages via the Microsoft Exchange Graphical User Interface – Subscriber Options, is controlled by the security scheme established for the client. Passwords changed from the GUI are displayed as asterisks (*) as an added security measure.

Users are encouraged to change their passwords often and can do so at any time from either the TUI or via the graphical user interfaces. Passwords can be set by the Administrator to expire every 1 to 999 days from the day they were last changed. A certain number of previous passwords (specified by the Administrator) may not be used. The user must change an expired password before being allowed to send or retrieve messages through the telephone user interface.




No one can access a user's password, as passwords are not displayed on any administration form. To allow for forgotten passwords, the Administrator has the ability to set a new default password that the user must reset upon login from either of the interfaces.

As an added security measure, the system locks the user’s mailbox after reaching the system parameter limit of consecutive unsuccessful login attempts, disconnecting the caller. This is tracked across multiple calls. The System Administrator must then unlock the mailbox for the user.

System Administration Security

A Modular Messaging solution offers multiple administration security features to protect the system.

Role-Based Access Control (RBAC) gives customers the ability to create administration accounts (logins) on the MSS based on customer-defined roles.

Customer-defined roles can be tailored to give each administrator only the access privileges that are needed to perform that administrator’s

job. When you set up an administrative role, you specify which web-administration pages the role can access and the access type. The access type can be "read and write" or "read only".

An Authentication, Authorization, and Accounting (AAA) server is an optional, customerprovided server that can be used to authenticate the credentials of administrators logging in to the MSS. The MSS can be configured to use one or two Remote Authentication Dial-In User Service (RADIUS), Lightweight Directory Access Protocol (LDAP), or Active Directory AAA servers to authenticate logins. For RADIUS servers, administration accounts on the MSS that will be authenticated by the AAA server, must also be defined on the AAA server.

In addition, information about administration activity on the MSS is always sent to logs on the MSS server. In addition to logging this information locally, for Release 3.1, the MSS can be configured to send logging information to an external, customer-provided server using the syslog protocol (RFC 3164). Only one syslog server can be administered, however, that server can be configured as a syslog relay server that forwards logging information to multiple syslog servers. Logging information stored on the MSS can be viewed using the web-administration interface.

The Mailbox Manager Log In window allows the Administrator to control access to Mailbox Manager with password protection. Upon initially starting Mailbox Manager, the log in window includes two choices: Support Technician and System Administrator. Mailbox Manager also allows the Administrator to create additional login names and passwords for which the Administrator can restrict access to certain types of records, known as Operators. Operators can modify their own passwords, after they log in to Mailbox Manager under their own subscriber names.

Voice Mail Domain Security

Access control lists can be edited to limit who has access to administration applications and tools in a voice mail domain. These lists define the following types of administration:

¾ System Administration – Members of the System Administration access control list can access and use all Modular Messaging Software administration applications and tools except subscriber administration tools. The default System Administration access control list has a single entry containing the account under which the first Messaging Application Server was installed.




¾ Subscriber Administration – Members of the Subscriber Administration access control list can use subscriber administration tools to enable subscribers to use Modular Messaging Software. The default list is empty.

Message Server Security

To enable the Messaging Application Server to log onto the Message Storage Server, one or more account names and credentials for the Message Storage Server must be configured.

Limiting Caller Transfer Capabilities

One of the biggest sources of toll fraud against businesses is through hackers transferring out of a voice mail system to an outside—usually long distance—number. The system protects against unauthorized transfers through call transfer controls that help prevent a caller from transferring to an outside line or number.

Toll restrictions can be implemented in the telephone system that restricts access to the lines used by the Modular Messaging server for call processing.

The Administrator can select the digits that callers are allowed to enter as the initial digit of an invalid mailbox number. This helps prevent toll fraud by prohibiting callers from obtaining an external line when dialing the initial digital of an invalid mailbox number.

The system can disconnect or transfer callers who make too many navigation errors, including entering invalid mailbox numbers (system wide parameter between 0 and 9).

7.2.0 Voice Mail Features 7.2.1 Forwarding The system must provide access for forwarded calls from:

• Customer telephone system

• Direct central office (Business or Centrex lines)

• 800 Service lines


Confirm support for each forwarding requirement.

Avaya Response:

Comply. Modular Messaging will accept calls forwarded from all three forwarded call scenarios.

7.2.2. Disconnect Detection The system should detect that a caller has hung up and immediately disconnect and restore the line to service.


Confirm support for this operation.

Avaya Response:

Comply.




7.2.3. Station Dialing In addition to the menu/route, callers may access an individual station either through the input of the extension number or the input of the called party's last name. A total of 2,000 names plus 100 extension numbers will be possible.


Confirm support for this operation.

Avaya Response:

Comply. Callers who know the extension number can enter it or access the Dial-By-Name feature. If less than ten names match the letters entered, those names are spoken in the form of a single digit menu and the caller chooses the correct recipient.

If greater than nine matches are found, the system will request the caller to enter more letters.

The total number of administrated users will be supported for station dialing.

7.2.4 Answer Announcement Individual, personalized announcements of 15-30 seconds for each mailbox user will be possible. A user's dictated answer message will only occupy the number of seconds dictated, with the remainder to be pooled so as to be available to:

• All other mailbox owners, and

• For message taking.

A system announcement of up to 30 seconds will be possible and also will be available in the event of switching system failure. It will be possible for the mailbox owner to input separate greetings for calls received internally or externally on the system. It will be possible for several individuals to share the same mailbox extension number. A caller reaching such a mailbox will be able to select between individual mailboxes.


Confirm support for these operations.

Avaya Response:

Partially comply. The proposed solution complies with all items specified above, except for programming a separate greeting for calls received internally or externally on the system. With the Call Handling feature, users can administer the system to differentiate between calls that reach a mailbox because an extension is busy or because there is no answer. Due to extensive research into providing the distinction between internal and external calls on the Modular Messaging telephone user interface, Avaya found that the technology has moved past this early voice mail feature. As more and more businesses have virtual and mobile workers, “external callers” can be employees as well. We found that providing the ability to distinguish between whether a line is busy (the greeting can indicate the user is in the office that way) or unanswered (the greeting can indicate the user is out for the day) provides information that is of greater use to all callers, whether fellow employees or business clients.




For the remaining requirements, we fully comply.

Personalized Greetings

Users can record and activate multiple personal mailbox greetings, which can be rerecorded and activated at any time. The Administrator sets the maximum length of greetings and prompts, which can be up to 30 seconds if desired. As recorded greetings only occupy the number of seconds recorded, this automatically frees disk space for other uses.

As an additional benefit, Modular Messaging uses industry standards to format and store messages, VoiceCon can determine which storage method is used. If GSM is selected, it provides “cell phone quality” encoded at an economical 13Kbps. As a result, a voice message that is one minute long will require approximately 95.2 KB.

If a personal greeting is not recorded, the system plays a standard greeting with the user’s recorded name and gives callers the option to leave a message.

Following are the types of greetings that can be made available to the user.

¾ Personal Greeting for All Calls

¾ Extended Absence Greeting – Callers cannot override listening to an Extended Absence greeting. After listening, the caller is offered the options of entering another extension, leaving a message, or returning to the Automated Attendant menu. The Extended Absence greeting overrides all Call Handling and Intercom Paging options. The user is reminded that an Extended Absence greeting is in place upon entering the mailbox and asked whether to turn the greeting off.

¾ Optional Greeting 1 and Optional Greeting 2 – Users can use these optional greetings to administer a call type, also known as Call Handling. With the Call Handling feature, users can administer the system to differentiate between calls that reach a mailbox because an extension is busy or because there is no answer. Users can decide which optional greeting to activate for Busy calls and No Answer calls.

In addition from either the graphical user interface or the telephone user interface, users can record a “Please Hold” prompt as well as their names.

System Failure Message

In the event of a messaging system failure, the proposed Avaya S8500C Server powered by Avaya Communication Manager can be administered to provide an announcement to callers who would normally be directed to the voice mail system.

Sharing Extensions

A caller application can be set up that allows up to nine users who share a telephone extension to receive their own voice messages in individual mailboxes. Callers choose from a single digit menu that plays the list of users associated with that extension. They then hear that user’s mailbox greeting and can leave a message. Users then access their individual password-protected mailboxes to retrieve their messages.

7.2.5 DTMF Signaling The system will be capable of receiving and generating standard DTMF tone signaling.





Confirm support for this feature.

Avaya Response:

Comply

7.2.6 Greeting Voice mail calls will be answered on the first ring and be time-and-date stamped.



Avaya Response:

Comply

7.2.7 Escape A caller reaching the voice mail system will have the ability to re-route to an extension by dialing up to five digits or the operator by dialing "0" before or after leaving a message. It will not be possible for a caller reconnected to the telephone system to be connected to the public network.



Avaya Response:

Comply. Callers can choose to enter a different extension or press “0” to reach an operator. For each voice mail domain, the Administrator can specify all the conditions for transferring a caller to an operator. A default number and an after-hours number can be set to which callers are transferred when they press “0” to access assistance. Rotary caller options can be set to either transfer the caller to an operator or to disconnect after no input is received. For callers experiencing difficulty navigating the system, a maximum number of errors can be set before transferring the caller to the operator. The Administrator can also designate a unique zero-out destination for each user’s mailbox. If a personal operator is not designated, callers are transferred to a default operator mailbox number.

Measures to prevent toll-fraud are discussed in response to 7.1.1. Human persuasion and/or misleading statements could make it possible for a caller to be placed in a position to reach an outside line.

7.2.8 Trunk Access It will be impossible for a caller passing through the attendant to reach an outside line.



Avaya Response:

Comply with clarification. Measures to prevent toll-fraud are discussed in response to 7.1.1. Following these guidelines will reduce the chances of a caller passing through the attendant to reach an outside line. Human persuasion and/or misleading statements could make it possible for a caller to be placed in a position to reach an outside line.




7.2.9 Distribution Lists The system will contain a minimum of 80 distribution lists of at least 25 names each plus "all broadcast."



Avaya Response:

Comply. Personal distribution lists, system distribution lists, and a broadcast to all features are all available.

Personal Distribution Lists

A personal distribution list (PDL) is a labeled collection of addresses that subscribers create and save for later use in order to quickly send a message to a group. Messages addressed to the mailing list are sent to each list members. Unlike administrator-created system lists, only the subscriber who created the PDL can view and use it. A PDL is not available for other subscribers to use. Modular Messaging supports up to 500 PDLs with up to 999 members (list entries) per subscriber, subject to system capacity, although no hard limit has been established for the product. Lists are not subscribers and do not count toward any purchasable limits.

Subscribers can address messages to PDLs from any of the following interfaces, applications, or clients:

¾ The Modular Messaging TUIs: Aria, AUDIX, and Serenade for Modular Messaging

¾ Modular Messaging Outlook Client

¾ Modular Messaging Lotus Notes Client

¾ Modular Messaging Web Client

¾ UCC Speech Access client

¾ A standards-based email client

System Distribution Lists – Enhanced-List Application

The Enhanced-List Application (ELA) is a messaging tool created and managed by the Administrator that greatly expands the capability of the Modular Messaging Message Storage Server to deliver a message to a large number of recipients. The system supports a maximum of 1,000 ELA lists, with up to 1,500 members per list. ELA can be nested to create larger lists. Modular Messaging ensures that if a target recipient is included on multiple lists that are referenced directly or indirectly, the recipient will receive only one copy of the message.

ELA associates one mailbox to a list of members. When a message is delivered into the list mailbox, known as the shadow mailbox, the ELA software distributes the message to the

members of the list. ELA members can be local or remote subscribers, and arbitrary email addresses, thus providing extreme flexibility. Ordinarily, recipients receive the message as if it were sent by the message originator. Recipients can reply to the person who originally sent the message and to all recipients of the original message. However, administrators can configure lists to block recipients from replying to ELA senders or recipient lists by placing the list in a communication that cannot receive messages.




An ELA mailbox is just like any other mailbox, allowing such operations as recording a name for the list, a greeting for the list, and allowing Call Answer messages to be distributed via ELA. Like any other mailbox, an ELA mailbox has a mailbox number, a numeric address, and a Modular Messaging email address. The actual addressing format used to send the message to the list mailbox can be any addressing format that the relevant interface (TUI or computer interface) supports. The Administrator can also administer a remote mailbox with the email address of a list, thus making Microsoft Exchange and IBM Lotus Domino lists available through Modular Messaging Message Storage Server configurations.

Broadcast

The Broadcast feature enables an Administrator to designate any Enhanced-List Application (ELA) to be a local broadcast list. When a message is received into an appropriately configured enhanced-list mailbox, the message is sent to all local subscribers and to all list members. The Administrator can also set up system-wide or enterprise-wide broadcast lists. Settings control which subscribers have message addressing privileges to ELA broadcast mailing lists.

The actual addressing format used to send a broadcast message can be any addressing format that the relevant interface (telephone user interface or PC interface) supports. While a new broadcast message does not activate Message Waiting Indication (MWI), Find Me, and Notify Me, it does activate Call Me, provided the broadcast message meets subscriberspecified criteria. If customers require MWI activation, the Administrator can create an ELA list with entries for each local subscriber in order to activate MWI.

The Modular Messaging telephone user interfaces announce broadcast messages as such. They present new broadcast messages before other messages and provide a summary count of broadcast messages after logging in. The Modular Messaging Web Client, Modular Messaging Outlook Client, and Modular Messaging Lotus Notes Client (all graphical user interfaces) provide visual indications for identifying broadcast messages.

7.2.10 Message Forwarding Messages may be forwarded to single or multiple destinations with or without introductory comments.



Avaya Response:

Comply

7.2.11 Audit Trail It will be possible for a user to designate a necessary written record of message destination, input time and receipt. This audit trail will be printed on the administrative console together with daily reports.



Avaya Response:

Comply. A record of messages will appear in the user’s Sent folder with time and date information. The Transaction Log Database provides a historical record of all activity in a mailbox. The Administrator can choose to print the log at any time. By adding the optional eHealth for Modular Messaging reporting application, this audit trail can be emailed as an Excel spreadsheet, viewed by the Administrator, or automatically printed daily.




7.2.12 Message Indication The receipt of a message in a mailbox will cause a message-waiting lamp or "stutter" dial tone upon lifting of the station handset to indicate a message-waiting condition.



Avaya Response:

Comply

7.2.13 Identification Code Users accessing the system will input a discrete six-digit identification code which will be positively validated prior to access to their mailbox. Identification codes may be changed by mailbox owner.



Avaya Response:

Comply. Please see the response to 7.1.1 for details regarding personal mailbox security and the use of identification codes (passwords).

7.2.14 Message Recovery The mailbox owner accessing the mailbox will be automatically told how many new messages have been received since last access and how many saved messages exist. Upon accessing the messages, the subscriber will have the choice of deleting, skipping or saving a message. Saved messages may only be deleted by the subscriber or by the system administrator.



Avaya Response:

The Modular Messaging telephone user interface provides the number of new broadcast messages, the number of new voice and fax messages, as well as the total number of saved messages stored.

From the graphical user interfaces, the subscriber can easily prioritize message review. Subscribers can also mark read messages as unread, thus moving messages from the Saved category to the "New" category.

Using either telephone or graphical user interfaces, subscribers can choose to save, skip, or delete messages, as well as forward or reply. Saved messages may only be deleted by the subscriber or the system administrator.




7.2.15 Message Reply A mailbox owner may respond to a message input by another system mailbox owner by simply depressing a single key.



Avaya Response:

Comply. After listening to a voice message, users press “8” to access the following reply options. Replies can be made with or without the original message attached, and can be made to only the message sender or all message recipients (which can be listened to before choosing to reply to all). If supported by integration, users can also call the sender.

7.2.16 Message Review It will be possible for a user to review and edit either an announcement or input a message.



Avaya Response:

Comply

7.2.17 User Controls A user accessing their mailbox must be capable of the following control functions:

1. Playback messages

2. Skip to next message

3. Cancel review

4. Replay last message

5. Replay faster or slower

6. Pause

7. Append information

8. Forward message (to mailbox or list)

9. Create new answer announcement

10. Increase play-back volume


Confirm support for this feature. Indicate if any function is not supported.

Avaya Response:

Comply. The proposed solution supports all of the above features.




7.2.18 System Management Console The system will be equipped with a CRT and printer to provide system management functions. The administrative programs and traffic information secured will be possible during system operation. Traffic reports will be available on customer demand or automatically on a pre-programmed basis in quarter, half or one hour time frames or daily and weekly. At a minimum, they will indicate the following:

• Storage space used for announcements or information mailboxes

• Storage space used for messages

• Maximum storage space used during the interval


Confirm support for this feature. Indicate if any requirement is not supported.

Avaya Response:

Partially comply; Avaya’s solution does not include a monitor (CRT) or printer they must be provided by VoiceCon.

The Message Storage Server is a Linux server; as such, separate storage indicators are not provided or needed to indicate the space utilized by announcement, information mailboxes, and messages. Linux server reports can be utilized to see the amount of overall storage used.

Modular Messaging provides detailed standard reports. Once generated, reports can be viewed onscreen and automatically sent to a printer, if desired; however, the printer is not provided as part of the solution. In addition, reports can be exported for archiving or additional manipulation purposes by using alternative tools. The export facility supports a number of popular spreadsheet, word processor, and data interchange formats. These include character separated values, comma separated values, Crystal Reports, data interchange format, Microsoft Excel, Lotus 1-2-3, record style (columns of values), Rich Text Format, tab-separated text, tab-separated values, ASCII text, and Word for Windows. An exported report file can also be attached to a message sent via a MAPI-enabled email system.

Following are the available standard reports for each server.

Avaya Messaging Application Server Reports

The Reporting Tool can be used by Administrators to generate predefined Messaging Application Server Reports for monitoring voice mail system usage, planning capacity, and tracking system security. The information is taken from the Transaction Database and generated for the voice mail domain. Some reports can also generate Messaging Application Server-specific or subscriber-specific information.




Using the Reporting Tool, Administrators can generate the following Messaging Application Server standard reports.

Hourly Records detailed information regarding the number of incoming Statistics and outgoing calls each hour in a specified time period. This

information is useful for monitoring incoming and outgoing call patterns for the voice mail domain.

Login Failures Records detailed information regarding unsuccessful mailbox logins due to an incorrect password or invalid mailbox number. A defined time period can be specified. This information is useful to help monitor system security for the voice mail domain.

Basic Metric Records statistics on messaging activity in the voice mail domain. It includes general information on telephone user interface usage and statistical information on subscriber telephone user interface logins. The contents of the Basic Metrics report are the core Key Performance Indicators for the voicemail system; hence this report also provides a general performance overview.

Port Statistics Records detailed incoming and outgoing call information for each port configured in the voice mail domain. The time period, ports, and MASs can be specified. This information is useful for monitoring port usage.

System Usage Records detailed call and messaging statistics for the voice mail domain. These include statistics regarding general call information, callers’ actions, incoming and outgoing call summaries, and message summaries. A defined time period can be specified. This report can be used to help monitor usage of the system for such things as the types of calls, the number of messages left, the number of fax calls, and the time ports were used for text-to-speech, and their disposition.

User Mailbox Statistics

Records detailed information regarding telephone calls and messages received by each mailbox in the voice mail domain. The time period and mailbox can be specified.

Avaya Message Storage Server Reports

The Avaya Message Storage Server collects information about system settings and attributes and information that depicts how the system is used, including data about features, subscribers, communities, data port loads, and remote messaging traffic. This information is displayed in real-time dynamic report pages, and in messaging traffic reports.

These reports reflect:

¾ Thick Client Access (Microsoft Outlook, Lotus Notes)

¾ Web Client Access

¾ UCC Speech Access, Access

¾ Message Networking Access




REPORT DESCRIPTION

Community Daily or Hourly Traffic Report

These reports show the total number of messages sent and received by each community. They also show the number of messages that were not sent or received by each community due to restrictions on sending during any day in the last 32-day period or any hour in the last seven days.

Feature Daily or Hourly Traffic Report

These reports show traffic information on a feature-by-feature basis. Features are divided into call answer features and messaging features.

Load Daily or Hourly Traffic Report

These reports show daily load traffic information for 1 to 32 days or hourly traffic information for the last 7 days. Traffic load refers to the message traffic and storage relative to established mailbox thresholds.

Network Load Daily or Hourly Traffic Report

These reports show network channel traffic one day at a time for up to 32 days or one hour at a time for any hours within the last 7 days. These reports can show any nodes that are exceeding specified threshold limits, the number of calls that went unanswered, the number of calls on each channel, and other channel traffic information.

Remote Message Daily or Monthly Traffic Report

These reports show information about traffic loads between a local messaging machine and a specified remote messaging machine.

Report of Classes of Service

This report shows information about the current classes-of-service.

7.2.19 Traffic Reports Traffic reports will be available on customer demand or automatically on a pre-programmed basis in quarter, half or one hour time frames or daily and weekly. At a minimum, they will indicate the following:

• Storage space used for announcements

• Total calls answered

• Total calls routed to station

• Total calls routed to default

• Total calls abandoned

• CCS use and call count by input


Confirm support for this feature. Indicate if any requirement is not supported

Avaya Response:

Comply with clarification; the first five requirements are provided by Avaya Modular Messaging for details of the reports please see the response to 7.2.18. Reports cannot be scheduled in incremental hours as requested.




The remaining stated requirement (CCS use and call count by input) will be provided by the S8500C Media Server with Communication Manager 5.0. This report can be scheduled in incremental hours as requested.

7.2.20 System Changeability It will be possible for the system administrator to add and/or delete mailboxes, change general recordings and perform other administrative duties while the system is in operation.


Confirm support for this feature

Avaya Response:

Comply

7.3.0 Networking VoiceCon plans on networking it new HQ messaging system to other VoiceCon locations equipped with messaging systems.

Avaya Response:

Read and understood

7.3.1 AMIS The proposed messaging system must support AMIS networking standards.


Confirm support for this networking capability.

Avaya Response:

Comply via the optional Avaya Message Networking system. Modular Messaging uses Internet standard networking protocols for message transport and directory updates. It supports the SMTP/MIME protocol for message transport and supports a Lightweight Directory Access Protocol (LDAP)-based protocol to communicate with other networked Modular Messaging systems. When networked with an optional Avaya Message Networking system, Modular Messaging servers can communicate via such protocols as AMIA-A and VPIM to other messaging servers that use those industry standard protocols, as well as to servers using Avaya proprietary network protocols.

7.3.2 Digital IP Networking The proposed messaging system should support VPIM networking standards.


Briefly describe digital networking capabilities of your proposed messaging system solution. Indicate if VPIM is supported.

Avaya Response:

Comply via the optional Avaya Message Networking system. The optional Avaya™ Message Networking system is a network integrator, allowing Modular Messaging Servers to communicate with other messaging servers that use supported industry standard and Avaya proprietary network protocols. This turnkey server-based solution connects individual voice or multimedia messaging systems.




Using a "store-and-forward" approach to networking, the Message Networking solution receives messages, performs the necessary protocol conversions, and delivers the message to one or multiple recipients on one or multiple messaging systems. The hub-and-spoke network topology, where each messaging system is directly linked through the Message Networking solution to every other messaging system, is simpler to manage and easier to expand than a conventional point-to-point topology.

Each messaging system requires only a single connection to the Message Networking solution, and new systems can be added easily without affecting existing servers. All message routing, protocol conversions, administrative functions, and management capabilities reside on the Message Networking solution, so network connections are less costly and easier to manage.

The Message Networking solution eliminates long-distance charges when customers use digital TCP/IP to transport messages over their wide area network (WAN). In addition, the Message Networking solution can network from 2 to 500 remote locations. It can expand messaging capabilities as customers' networks grow.

Message Networking provides transcoding between diverse digital voice encoding formats and converts analog-to-digital and digital-to-analog. The Message Networking solution supports TCP/IP (for INTUITY™ AUDIX®, Octel® 200/300, Octel® 250/350, and non-Avaya messaging servers that are VPIM V2-compatible), AMIS-analog, and Octel-analog protocols. To design and implement the Avaya Message Networking solution, customers have the option to administer their own end points (thereby reducing their ongoing implementation costs) or choose one of the specially designed Professional Services offers. With one of these offers, the Professional Services Organization designs and/or implements the Message Networking for our customers. Please note that if your network includes Interchange as one of the end-nodes, a Professional Services offer is required to implement the Message Networking solution.

Through various methodologies, the Avaya Message Networking server maintains a complete and current enterprise-wide directory of all users in the network. Up to 500,000 networked users are supported. The Message Network server can provide all or selected views of the enterprise directory to the Modular Messaging server, complete with ASCII names and network addresses. The directory updates can also include up to 120,000 spoken names. These directory updates ensure that users can address by name and receive name confirmation while addressing messages to networked users.

In addition to the above-mentioned features, the Message Networking solution also offers:

¾ LDAP-based Directory Update Support for a Modular Messaging solution

¾ Web-based Administration Interface

¾ Test Connect Tools

¾ On-line Help

¾ Call Detail Recording




7.4 Integrated Messaging Application VoiceCon requires that its voice messaging system be fully integrated with its MS Exchange messaging solution to provide unified messaging access from any system terminal/client.


Confirm that the proposed voice messaging system supports this requirement. Briefly describe how the proposed voice messaging system can be integrated with VoiceCon’s text messaging system, based on a MS Exchange server running the latest version of Exchange, to provide unified messaging system functionality. Station users must be able to view and access all messages (voice, text, fax) from their PC display monitor. Email text messages must be accessible from a telephone using text-to-speech conversion.

Avaya Response:

Avaya is proposing an Avaya Modular Messaging solution with an Avaya Message Storage Server. To provide unified messaging capabilities, graphical user interface (GUI) clients can be made available to the user to provide access to voice, fax, and optionally email messages from a PC. These applications provide a visual interface to perform various operations such as accessing and sending messages; managing messages; configuring mailboxes; and maintaining mailboxes, rules, greetings, and TUI and GUI preferences.

The following GUIs apply to Modular Messaging with a Message Storage Server.

¾ Avaya Modular Messaging Microsoft Outlook Client: This client application creates a new Inbox for Modular Messaging messages, separate from the corporate email inbox. Users can access, send, and manage messages from this Inbox within the Microsoft Outlook email application.

¾ Avaya Modular Messaging Web Client: This Avaya client application allows users to access, send, and manage voice, text, fax, and corporate email messages from a Web browser. While it cannot be used for mailbox administration functions, users can still do this from the TUIs or from Subscriber Options.

¾ Standards-based Clients: Users can receive, send, and manage messages from a standards-based email client that supports either IMAP4 or POP3, although IMAP4 is preferred for its enhanced capabilities. While this method cannot be used for mailbox administration functions, users can still do mailbox administration from the TUIs or from Subscriber Options.

¾ Subscriber Options: This application provides users the capability to modify their mailbox settings from a desktop PC. Subscriber Options integrates with the Microsoft Outlook or the IBM Lotus Notes email client. It can also work as a stand-alone application.

¾ Web Subscriber Options: This application provides users the capability to modify their mailbox settings from a Web browser. Web Subscriber Options provides all of the functionality that Subscriber Options provides. Authentic users must enter their mailbox numbers and numeric passwords to enter Web Subscriber Options. Users can also log in to this application using a Quick Logon from the Modular Messaging Web Client.

The solution also includes Avaya one-X Speech, which provides the following capabilities via integration with Outlook.




Access to & Control of the information already in place!

Messages • Read • Reply • Forward • List • Filter • Send to a

contact

Contacts • Call a contact • Send messages to a

contact • Get phone numbers • Get email addresses • Get mailing address

Calendars • Review Appointments • Create Appointments • Read, reply to and forward

meeting requests • Accept or decline meeting

requests • Set Reminders

Tasks • Read • List • Filter • Reply to

request

Calling • Call any contact • Dial any number • Reply to voice

message with a call • “Single Number”

Find-me/Follow-me (Hide-me)

• Add up to 23 parties toconference

• Speak commands duringconference

• Leave and re-enter conference • Continue two-party call after

conference

Conference Calls

44 Avaya – Proprietary Use pursuant to Company instructions



Part 1 - Section 8 - Unified Communications

Section 8: Unified Communications VoiceCon desires to conduct a departmental trial of a Unified Communications (UC) solution prior to implementing it across the entire organization. The initial trial is required to support the personal productivity needs of 50 station users situated at the CHQ facility.

VoiceCon defines a UC solution as one that supports the following features and capabilities:

• Station user programmed call screening, coverage, and routing

• Presence management & control

• Conferencing tools for audio, data, and video communications

• Collaboration tools for desktop screen sharing, management and editing


Confirm your proposed IPTS communications solution includes a UC option that supports the requirements and capabilities outlined in the following RFP sections, and provide a brief overview of your UC offer.

Avaya Response:

Comply; Avaya Unified Communications solutions are integrated, multi-vendor business communications applications, systems and services that work together in a reliable and secure fashion. They are able to work through a seamless user interface that is independent of device. The result is a superior, seamless user experience across all enterprise communication solutions regardless of location, network, or device.

Avaya’s Unified Communications Solutions include:

For Unified Access:

One-X Desktop- (SIP Softphone)

Softphone client to control your phone calls and telephony features from your PC. Allows workers to use their desk phone in tandem with their PC while at the office, direct the voice path to another phone to support virtual or work-at-home, or voice over IP

IP Softphone with Microsoft Office Communicator or IBM Lotus Sametime

Integrated telephony with Microsoft Office Communicator (MOC) allows users to click-to-call or click-to-video from the ‘buddy’ list, an email, or a smart tag. They can then access all Avaya Communication Manager features, gain integrated desktop video, and click-to-call from Internet Explorer.

One-X Portal

A web-based software solution that provides users with control of telephony, conferencing, messaging, mobility administration, and access to corporate and personal contact information. The solution complements the use of the desk phone while in the office, supports remote workers from any location, and is an ideal component of a business continuity strategy in that it can be used by any employee from any internet connection should they be unable to get to an office location in the event of a business interruption.




One-X Mobile

Makes the mobile phone an extension of the desk phone - calls simultaneously ring on both and can be answered on either. Seamlessly shift from desk to mobile phone or mobile to desk phone while a call is in progress. Calls placed from your mobile phone go through the enterprise communication system to take advantage of least cost routing, management, reporting, and single number identity. Maintain a single corporate voice mailbox and access to LDAP corporate directory and a consolidated call log from your mobile phone.

one-X Speech (Unified Communication Center (UCC) Speech Access)

Hands-free/eyes-free/speaker-independent speech access to voice mail, e-mail, calendars, directories, calling, and conferencing.

For Conferencing & Collaboration:

Meeting Exchange

Audio conferencing solution that combines reservation-less, attended, scheduled meet-me, event-based, capabilities; sub conferencing, dial out, blast dial, recording, billing and reporting features. Supports 24 to 8000 users in a single system and multiple systems can be deployed as needed. Supports both VoIP SIP and TDM protocols, and can be aligned with gateways (Communication Manager for example) to support H.323.

For Adobe, IBM, Microsoft

Use 3rd party web conferencing clients such as IBM Sametime, Microsoft Live Meeting, or Adobe Acrobat Connect Professional to see conference presence and who is speaking. Control audio components such as mute/un-mute lines, add participants, etc.

Avaya Web Conferencing

Browser-based collaboration solution that allows users to hold productive virtual meetings by sharing applications or the desktop, reviewing presentations, and white boarding. Integrating with Meeting Exchange provides audio conference controls embedded in the meeting interface, allowing users to identify speakers, mute individuals, dial out to bring new people into the call, or record the audio and Web proceedings.

Video Telephony

Avaya makes video as easy and full-featured as a phone call by adding video to its IP telephony capabilities. Options range from desktop video using Avaya IP Softphone, group conference room video, multi-point video, Telepresence, and high definition video. Polycom or Tandberg video endpoints can be added to the solution. Avaya video solutions can be used for a range of applications including distance learning, telemedicine, customer service, travel reduction, or everyday workforce productivity.




8.0.1 UC System Integration Vendor Response Requirements:

How do the proposed IPTS and voice messaging systems physically and logically integrate with the proposed UC solution?

Avaya Response:

IP SOFTPHONE

The IP Softphone is a TAPI-compliant application that functions over the IP network in conjunction with Avaya Communication Manager. The Avaya IP Softphone is connected either directly using a network interface card (NIC), by a dial-up modem with a point-topoint (PPP) account, or by an internet connection. The Avaya IP Softphone Release 6.X will run on Microsoft Windows 2000 or Windows XP software. Avaya IP SoftPhones include the Remote Help Desk Support offer, which can help to simplify installation and ease deployment.

MICROSOFT OFFICE COMMUNICATOR (MOC)

Avaya offers two telephony integrations for Avaya Communication Manager with Microsoft Live Communications Server 2005 (LCS) and Microsoft Office Communicator (MOC). Both solutions provide integrated telephony and instant messaging presence, and click-to-call capabilities from Office Communicator and Microsoft Smart Tags. One offer is Applications Enablement Services 4.0 (AES) which is a centralized server solution requiring nothing to be installed on the user’s PC. The other offer is IP Softphone 6.0 which is a Windows client solution installed on a user’s computer. IP Softphone adds an Avaya tab to Office Communicator, providing direct access to advanced Avaya Communication Manager features. IP Softphone also adds additional capabilities to manage voice and video communications.

Microsoft Office Communicator integrates with Meeting Exchange Conferencing as well.

ONE-X DESKTOP (SIP Softphone)

Avaya one-X Desktop Edition supports SIP capabilities as per IETF RFC 3261. It contains a set of components and drivers (DLL) including:

1. Registration and connectivity functions

2. Audio (VoIP) functions

3. Call Control

4. Presence and Instant Messaging

5. Search Components for LDAP, Microsoft Outlook, Lotus Notes

6. Local User information including contact lists, speed dial numbers and call history logs.

ONE-X PORTAL

one-X Portal is a web based interface allowing telephony access (telephony, messaging, conferencing, and mobile communications) through a secure, VPN SSL web client.




Client Desktop - PC

¾ Pentium III 800mz or better, 512M RAM, 1.5 GB of free disk space, CD ROM or DVD ROM disk drive, keyboard, mouse, microphone, speaker and sound card, 10 MBS NIC or better

¾ Microsoft XP SP1 SP2 or better, Microsoft Internet Explorer 6.0 or greater or Mozilla Firefox 1.5 or greater

Client Desktop - Mac

¾ 1.33 GHZ CPU, 512 RAM

¾ MAC OS 10 with Safari 2.0 or Mozilla Firefox 1.52 or better

Avaya one-X Portal Server

¾ Dell 1950 Two (2) Dual-core processors @ 1.86 GHz or equivalent, 4GB RAM, 80 GB drive, 100 MBS NIC, DVD/CD Combo Drive

¾ RedHat Enterprise Linux ES 4.0 Update 4, WebSphere 6.0.2.x with Web Portal Plug-in 5.1.x.x

Minimum Server Requirements

¾ Avaya Communication Manager 3.1

¾ Modular Messaging 3.0 SP2 with the Avaya Message Storage Server

¾ Meeting Exchange Enterprise 4.1

¾ Application Enablement Services 3.1

¾ Microsoft Active Directory Server 2003 SP1

ONE-X MOBILE

The Avaya one-X Mobile platform provides integrated software applications for mobile telephony devices. These applications provide simple user access to telephony and unified communications. Avaya one-X Mobile is specifically designed to improve and simplify user experience and access by providing graphical interfaces on popular operating systems for mobile phones.

Avaya one-X Mobile supports a broad number of devices across both its telephony and UC application integration deployment models. Today, the telephony integration deployment supports mobile devices using Symbian and Windows Mobile 5 mobile operating systems. When using the unified communication application integration deployment model, Avaya one-X Mobile supports mobile devices which operate in Palm, RIM, J2ME or WAP environments. For a full list of supported devices by Avaya one-X Mobile client type please go to support.avaya.com for the most up to date listing.




Platform Overviews

Unified Communication Application Integration deployment - RIM, Palm, Java

Telephony Integration deployment - Symbian, Windows Mobile 5

Avaya one-X Mobile provides a robust set of standard features across all deployment options. Features such as single number access, single voicemail box, easy mobile switcher and dialing out via the PBX provide a the foundational elements critical to any mobile worker. In an UC application integration, additional features are made available through a deeper integration with key applications, such as voicemail and corporate directory to further enhance the mobile workers productivity and overall experience. The server components of this deployment option provide the conduit back to these applications and deliver features such as Visual voicemail, corporate directory search and click to call as well as enhanced call routing features directly to the mobile user. These features further enhance the mobile worker's responsiveness and productivity while helping to control the overall mobile costs of the enterprise. The architecture and deployment of the required server hardware will be dependent on the enterprise network and security requirements.

Avaya enterprise mobility solutions help organizations extend the Avaya suite of IP enterprise applications over any network to workers on the move anywhere around the world.

In an telephony integration deployment, Avaya brings an integral mobility feature to the partnership via Avaya Communication Manager. Avaya Communications Manager, part of the company's IP-based MultiVantage Communications Applications suite, greatly enhances the productivity and experience of S60, S80, and Windows Mobile 5 based mobile phone users. The software enables the extension of popular office phone features, including call conferencing, transfer and abbreviated dialing, directly supported mobile phones, via an easy to use, integrated menu feature.

AVAYA WEB CONFERENCING

Avaya Web Conferencing is a software application that is installed on a customer premise server, and can be accessed by licensed users. Users must download a Java client on their PC, but otherwise, the application can be easily accessed via an Internet browser and any connection speed.




Hardware:

Server Specifications Choice of server:

¾ Dell 1950 PE server

¾ HP ProLiant DL 380 server

OS - Windows 2003 Server with IIS 6

Browser Support Windows

¾ Microsoft IE 5, 5.5, and 6

¾ Netscape 6.1 and 6.2

¾ Netscape Navigator 4.5, 4.61, and 4.7x

SPARC Solaris

¾ Microsoft IE 5 Service Pack 1

Netscape Navigator 4.7

¾ SGI IRIX

¾ Netscape 4.77

HP-UX

¾ Netscape 4.77

¾ Linux (Red Hat Linux 6.0 and Caldera Open Linux 2.2)

¾ Netscape 4.7

Macintosh

¾ Microsoft IE 4.5 and 5

Netscape 4.7

Software:

Client Side Software Sun Java 1.4.2.06 / MS JVM Requirements

IP VIDEO

Leveraging Avaya Communication Manager as a converged gatekeeper for both voice and video networks allows direct connection between Communication Manager, the Polycom Media Gateway Controller (MGC) multipoint control unit, and the video endpoints. This makes it possible to deploy a unified network management system for voice and video, including: common signaling, a standardized dial plan, integrated directories, and common authentication and authorization procedures. The Solution allows Polycom video endpoints to be automatically registered with Avaya Communication Manager for seamless initiation of the video conference. New video capabilities of Avaya Communication Manager V.3.0.1 and higher include: support for standards based endpoint registration (H.323), secure,




automatic authentication of Polycom VSX series endpoints and Avaya IP phone and Softphone along with new gatekeeper capabilities.

Software:

Avaya IP Softphone R 5.2 + Avaya Integrator for Polycom Video R 2.0

¾ Integrated with Avaya Manager as gatekeeper

Communication

¾ Supports hold, transfer, and coverage via Communication Manager

¾ Supports IP Softphone, VSX, and MGC

¾ Supports send only and receivevideo, as well as video mute

only for

¾ Supports USB Web Cams

PC Specifications for Support of IP Softphone with Video

Operating System Support ¾ Windows 2000, Service Pack 3 or higher

¾ Windows XP, Service Pack 1 or 2

Processor ¾ 1 GHz for USB camera

¾ 1.5 GHz to invoke H.264 codec (USB cameras only)

¾ USB Support minimum USB 1.1 or 2.0

¾ RAM >512 MB

¾ Hard Drive Space >50 MB

¾ Microsoft DirectX Software v9.0b or later




Hardware

Avaya Communication ¾ S8700 or S8500 server (supports 1000 video calls) Manager ¾ S8300 server (supports 75 video calls)

¾ Communication Manager H.323 trunk to a third party gatekeeper

Additional Hardware ¾ Polycom VSX Room Systems:

¾ V500

¾ VSX 3000

¾ VSX 5000

¾ VSX 7000s

¾ VSX 7000e

¾ VSX 8000

¾ Polycom MGC Multipoint Control Units:

¾ MGC 25

¾ MGC 50

¾ MGC 100

8.0.2 Presence Management Vendor Response Requirement:

Confirm that presence management is fully integrated into the UC solution. Include a brief description of presence features and functions included with the offering. Indicate if presence management requires a dedicated server or gateway with middleware.

Avaya Response:

Comply.

Avaya IP Softphone:

IP Softphone offers simple Graphical User Interfaces (GUIs) and integrates well with Microsoft desktop applications such as Outlook and Internet Explorer, enabling click to dial and screen pops. There are options to add Instant Messaging and Presence as well as videoconferencing for enhanced collaboration capabilities. IP Softphone links with your desk phone so that calls to you will ring at your desk phone and PC.

Avaya IP Softphone integrates a flexible IP telephone client with a SIP/SIMPLE-based Instant Messaging (IM) client. The IM client incorporates a contact list of other IP Softphone users and makes both phone and IM presence visible to other users. It is simple to toggle between the Softphone and IM applications. The IM and presence capabilities require registering with the Avaya SIP Enablement Service (SES) platform, which is available separately.




Avaya Communication Manager with Microsoft Live Communications Server 2005 (LCS) and Microsoft Office Communicator (MOC)

Avaya offers two telephony integrations for Avaya Communication Manager with Microsoft Live Communications Server 2005 (LCS) and Microsoft Office Communicator (MOC) 2007. Both solutions provide integrated telephony and instant messaging presence.

One offer is Applications Enablement Services 4.0 (AES) which is a centralized server solution requiring nothing to be installed on the user’s PC. This integration and click-tocommunicate drives greater efficiencies and productivity improvements for the typical enterprise telephony user by enabling numerous desktop telephony features, which eliminate manual dialing while driving all communication operations to a single desktop interface (i.e., Microsoft OC client). The telephony capabilities provide presence Status (On/Off Hook status integrated with MOC IM client).

The other offer is IP Softphone 6.0 which is a Windows client solution installed on a user’s computer. IP Softphone appears as a second tab in Microsoft Office communicator, allowing use of the Softphone in a familiar interface. This enables conversations using IM and providing presence awareness.

Avaya one-X™ Desktop (SIP Softphone)

Avaya one-X™ Desktop Edition transforms Windows-based PCs into intelligent SIP-based collaboration endpoints. Users are able to make, receive and handle voice calls, send instant messages to one another and track each others' presence using an intuitive graphical user interface. Avaya one-X Desktop Edition leverages the industry-standard nature of SIP and operates in three modes; with Avaya SIP Enablement Services as part of a complete Avaya SIP solution; with 3rd-party SIP Proxy Servers and endpoints that are IETF RFC 3261 compliant; and in peer-to-peer mode with no SIP infrastructure.

8.0.3 Instant Messaging Vendor Response Requirement:

Confirm that Instant Messaging (IM) is fully integrated into the UC solution. Include a brief description of IM features and capabilities, and also indicate if the integrated IM offer is interoperable with public IM services? Is IM recording supported?

Avaya Response:

Comply.

IP Softphone

Avaya IP Softphone integrates a flexible IP telephone client with a SIP/SIMPLE-based Instant Messaging (IM) client. The IM client incorporates a contact list of other IP Softphone users and makes both phone and IM presence visible to other users. It is simple to toggle between the Softphone and IM applications. The IM and presence capabilities require registering with the Avaya SIP Enablement Service (SES) platform, which is available separately.

IM does not interoperate with public IM services.




The Call History log stores information on incoming calls, outgoing calls, incoming Instant Messages, and outgoing Instant Messages by automatically generating a call record or oneline summary of each call/instant message. You can add or modify notes in the record, delete the call record, dial the selected number from the record, or send an Instant

one-X Desktop (SIP Softphone)

Avaya one-X™ Desktop Edition transforms Windows-based PCs into intelligent SIP-based collaboration endpoints. Users are able to make, receive and handle voice calls, send instant messages to one another and track each others' presence using an intuitive graphical user interface.

IM does not interoperate with public IM services.

The Call History log stores information on incoming calls, outgoing calls, incoming Instant Messages, and outgoing Instant Messages by automatically generating a call record or oneline summary of each call/instant message. You can add or modify notes in the record, delete the call record, dial the selected number from the record, or send an Instant Message.

Avaya Communication Manager with Microsoft Live Communications Server 2005 (LCS) and Microsoft Office Communicator (MOC)

Avaya offers two telephony integrations for Avaya Communication Manager with Microsoft Live Communications Server 2005 (LCS) and Microsoft Office Communicator (MOC). Both solutions provide integrated telephony and instant messaging presence.

When selecting the IP Softphone offer, instant messaging presence is available, but does not interoperate with public IM services. There is a Call History log that stores information on incoming calls, outgoing calls, incoming Instant Messages, and outgoing Instant Messages by automatically generating a call record or one-line summary of each call/instant message. You can add or modify notes in the record, delete the call record, dial the selected number from the record, or send an Instant Message.

8.0.4 UC Clients Vendor Response Requirement: What desktop PC client communications protocols are supported by the UC solution? For example, is a SIP-based client, only, supported or required? Identify the PC’s technical specifications required to support the UC client. Provide as an attachment a graphical illustration (PPT format, only) of a typical UC screen shot that is representative of your UC offering.5

Avaya Response:

Avaya provides an IP H.323 (Avaya IP Softphone) and SIP based (Avaya one-X™ Desktop) softphone solutions.

Avaya IP Softphone (H.323)

Avaya IP Softphone offers a tight integration with existing communication applications such as Microsoft Office Communicator, Lotus Sametime, and Citrix environments. This integration with existing communication tools offers smaller learning curves and a true Unified Communications experience.

5 Replaced by Allan via e-mail 11/6/2007




Avaya IP Softphone 6.0 offers additional integration with other communication applications such as Microsoft Outlook Communicator, Lotus Sametime, Citrix Presentation Server and with Service Pack 1, and can now support the Microsoft Vista Business and Ultimate Editions in a standalone configuration. Additionally, IP Softphone R 6.0 offers native support for new Communication Manager 5.0 features: 9600 series phones, signal channel encryption, and 13-digit dialing support.

The IP Softphone telephone client can be configured for VoIP (Roadwarrior) mode, Dual Connect (Telecommuter) mode or Shared Control mode. The user is empowered to establish the mode that best suites their current environment.

¾ VoIP (Roadwarrior) mode is suitable for users who have a single, broadband connection from their home office or remote location. In VoIP mode, the audio and call control signaling are run across the IP network for a "pure voice over IP" configuration, and offers a great amount of flexibility due to the ubiquity of IP networking. Voice communication is generally done via a headset connected to the PC.

¾ Dual Connect (Telecommuter) mode is ideally suited for users working from a remote office with a phone line. With this option, call control signaling is maintained and delivered across the IP network, but the voice is delivered across a phone line to either a public switched telephone network or digital line to help ensure toll-quality voice. This capability can be extended to any touch-tone phone or cellular phone.

¾ Shared Control mode enables the control of Avaya IP telephones or Avaya digital (DCP) telephones.

¾ The Avaya IP telephone configuration enables users to log into and control their Avaya IP telephone from the Avaya IP Softphone. Users can speak and listen through their telephone, but unlike the Multi-Phone configuration, users can make and handle calls from both the Avaya IP Softphone interface and the IP telephone. This feature can help improve productivity by integrating the IP Softphone capabilities and Personal Information Managers, such as Microsoft Outlook, with the IP Telephone. The Avaya IP Telephone configuration is supported on the Avaya 4600 or 9600 series IP Telephones.

¾ The Avaya DCP Telephone configuration enables users to log into and control their traditional digital telephone from the Avaya IP Softphone. This capability is supported on Avaya 2400 and 6400 series phones.




PC Minimum Requirements

IBM PC-compatible equipment with the following:

¾ IBM Compatible Personal Computer with the 800 MHz processor (1 GHz recommended for Road Warrior and Video).

¾ NOTE: Touch-screen systems have not been certified to work with Avaya IP Softphone.

¾ Hard disk with at least 50 MB of space available.

¾ RAM Minimum Requirements:

Operating System Road Warrior Telecommuter, Shared Control

¾ Windows XP (32-bit) 256 MB 128 MB

¾ Windows 2000 128 MB 64 MB

¾ Windows Vista (32-bit) 1 GB 512 MB

The minimum PC requirements assume IP Softphone is the only application running on the PC. In reality, most PCs will be running one or more other applications concurrently with IP Softphone. If you have not yet purchased a PC to run IP Softphone, select a PC that exceeds the minimum requirements. How far your PC should exceed these requirements depends on how memory intensive the applications are that you typically use. If you frequently use multiple memory intensive applications concurrently (such as a web browser), you should select a PC that has a fast processor and enough RAM.

¾ For the option of ringing on the PC, a sound device.

¾ For Road Warrior configuration only, a sound device that supports full-duplex operation (both parties can talk and hear each other at the same time).

¾ For Road Warrior configuration only, a speaker/headset, and a microphone.

¾ Network Interface Card (NIC) for local area network (LAN) connectivity and/or a modem (28.8 Kbps or faster) for dial-up networking.

¾ CD-ROM drive (if installing from CD).

¾ Microsoft Internet Explorer 5.5 or higher to view the online help and to support the ability to dial numbers from web pages.

¾ One of the following operating systems:

◊ Microsoft Windows XP Home or Professional Service Pack 2

◊ Microsoft Windows 2000 Professional with Service Pack 3 or higher

◊ Microsoft Windows Vista Enterprise or Ultimate

One-X Desktop Edition (SIP Softphone)

Avaya one-X™ Desktop Edition transforms Windows-based PCs into intelligent SIP-based collaboration endpoints. Users are able to make, receive and handle voice calls, send instant messages to one another and track each others' presence using an intuitive graphical user interface. Avaya one-X Desktop Edition leverages the industry-standard nature of SIP and operates in three modes; with Avaya SIP Enablement Services as part of a complete Avaya SIP solution; with 3rd-party SIP Proxy Servers and endpoints that are IETF RFC 3261 compliant; and in peer-to-peer mode with no SIP infrastructure.




Avaya one-X™ Desktop Edition is a flexible SIP endpoint that uses SIP for telephony, enterprise instant messaging and presence. It supports a rich suite of SIP-based telephony features and can be integrated with Microsoft Outlook, Lotus Notes, LDAP and Microsoft Internet Explorer. It can be used in any of the following three modes of operation.

1. Peer-to-peer mode with no SIP infrastructure.

This mode of operation requires no SIP proxy or registration server. It is a free-ware version requiring no purchased user license. It supports a basic set of telephony and instant messaging features between Avaya one-X Desktop Edition users. One limitation of this implementation is that there is no name resolution - that is, users must know the called party's IP address to contact them.

2. Industry-standard SIP endpoint with 3rd party SIP infrastructure.

In this environment, Avaya one-X Desktop Edition interoperates with 3rd party SIP proxy, registrar and endpoints. In this case, one-X Desktop Edition will implement the complement of standards-based SIP features (SIPPING 19) that the rest of the infrastructure supports. Thus the SIP Softphone can augment existing, standards-compliant SIP installations.

3. Endpoint of Avaya SIP Solution with Avaya SIP Enablement Services and Avaya Communication Manager.

This implementation leverages the Personal Profile Manager capability of Avaya SIP Enablement Services to provide a network-based contact store and access control lists for Avaya one-X Desktop Edition users. It also provides server-based presence, message-waiting indication and enhanced secure communications between Avaya one-X Desktop Edition and SIP Enablement Services. With the introduction of Avaya one-X Desktop Edition 2.1, a large complement of advanced telephony features, including click to conference, all forwarding, whisper page, malicious call trace, call park, and call pickup, are now available.

PC Requirements

Your PC must meet the following minimum requirements to run Avaya one-X Desktop Edition R2.1 Service Pack 2:

¾ IBM PC or compatible PC with an Intel Pentium III 800 MHz or compatible processor.

¾ A hard disk with at least 50 MB of space available. Additional space will be required (over time) to store entries in the contact directory database and in the call history database.

¾ One of the following operating systems:

◊ Microsoft Windows XP Professional with Service Pack 4

◊ Microsoft Windows XP Home with Service Pack 2

◊ Microsoft Windows 2000 Professional with Service Pack 2

¾ 512 MB of RAM for Microsoft Windows XP Professional

¾ 512 MB of RAM for Microsoft Windows XP Home

¾ 256 MB of RAM for Microsoft Windows 2000 Professional.




¾ One of the following network devices:

◊ a network interface card for local area network connectivity or wide area network connectivity

◊ a network interface card and a high-speed modem (cable, xDSL, or ISDN)

◊ an analog modem (28.8 Kbps or higher) for dial-up networking connections

¾ A Microsoft Windows-compatible video graphics adapter that supports 16-bit color and a video resolution of 1024x768 (or better)

¾ A Microsoft Windows-compatible pointing device (such as a mouse or touchpad)

¾ A sound card, speaker/headset, and microphone. The sound card must support full-duplex operation. A list of recommended sound devices and headsets is available at www.avaya.com/support.

¾ The following additional software:

◊ Microsoft Internet Explorer 6.0 or later

◊ Microsoft Windows .NET Framework 1.1 required (available from Microsoft). (Microsoft Windows .NET Framework 2.0 may also be installed in addition to the required 1.1 version.)

◊ Microsoft Outlook 2000 SR-1, Microsoft Outlook 2002, or Microsoft Outlook 2003 (if you want to search for Microsoft Outlook contacts from Avaya one-X Desktop Edition)

◊ Lotus Notes Release 5 or Release 6 (if you want to search for Lotus Notes contacts from Avaya one-X Desktop Edition)

If you want to use a Bluetooth headset with Avaya one-X Desktop Edition, you will need a Bluetooth adapter with the appropriate Bluetooth stack:

¾ Widcomm Bluetooth stack version 1.4 or later

¾ Toshiba Bluetooth stack version 4.01 or later

Please Refer to Avaya Appendix 7 for additional information and screen shots.

8.0.5 Client Control Vendor Response Requirement:

Can the PC client be used to access and implement IPTS telephony features for control of the desktop telephone instrument?

Avaya Response:

Yes please refer to 8.0.4 for additional information.




8.0.5.1 Call Routing Control Call coverage and routing control should include, at minimum, the following programming parameters:

• Caller priority level;

• Current station status

• Available route-to devices (desktop telephone instrument, mobile teleworking client, wireless handset, cellular handset, et al)

• Time-of-day/day-of-week


Describe how the client GUI is used to screen, manage and route calls on a customized basis. Confirm that the above programming parameters are supported as part of the call routing control mechanism.

Avaya Response:

Avaya provides mobility options for an environment supporting various levels of call coverage and routing control for mobile users. Employee mobile requirements can differ by their department and role within the organization.

For this reason Avaya provide flexible options, from as simple as a bridged appearance of the user office phone to their cellular device to a full featured PDA providing GUI control of Avaya Communication Manager, Visual Voice Mail with Modular Messaging and much more. The capabilities are all part of Avaya’s one-X Mobile family. In addition to the one-X Mobile family capabilities there are also call routing control options available in Modular Messaging and one-X Speech. Modular Messaging provides Find Me and one-x Speech provides Reach Me.

Avaya Extension to Cellular

Extension to Cellular bridges a mobile device to a desk phone providing true one business number access. The software based solution provides the foundation for the Avaya one-X Mobile family of clients and can be easily deployed to any mobile device though a simple activation of an Extension to Cellular license on the supported versions of Avaya Communication Manager.

Avaya one-X Mobile

Avaya one-X Mobile builds on the functionality of Extension to Cellular by providing a family of mobile clients designed to further enhance the productivity of enterprise mobile workers.

The family of Avaya one-X Mobile clients provides an intuitive graphical user interface to provide quick access to office phone functionality.

¾ Caller priority level;

Designate specific Phonebook contacts as VIPs for use in conjunction with the Block feature. When the Block feature is set to block non-VIP callers, a VIP can ring through while all other callers are sent directly to your office voicemail.

The Block feature, when set to Non-VIPs, allows VIP callers to ring through. All other callers are sent directly to your office voicemail

¾ Current station status

N/A




¾ Available route-to devices (desktop telephone instrument, mobile teleworking client, wireless handset, cellular handset, et al)

The clients support all major mobile operating systems (RIM, Palm, Symbian, Windows Mobile and J2ME) and devices ranging from high end smart phones such as the Apple iPhone to lower end feature phones providing support for most enterprise environments, independent of the mobile devices deployed or wireless networks used.

◊ The Avaya one-X™ Mobile family consists of the following clients:

◊ Avaya one-X™ Mobile for Symbian

◊ Avaya one-X™ Mobile for Symbian Dual Mode

◊ Avaya one-X™ Mobile for Windows Mobile

◊ Avaya one-X™ Mobile for RIM

◊ Avaya one-X™ Mobile for Palm

◊ Avaya one-X™ Mobile for Java

◊ Avaya one-X™ Mobile for iPhone *

¾ Time-of-day/day-of-week

Automatic by Schedule manages your incoming calls according to a defined schedule. When Automatic by Schedule is selected as a Send Calls destination, no other destinations can be selected.

There is additional information in section 8.0.6 where Mobile Clients are discussed in detail.

Modular Messaging Find Me

Find Me is a feature that redirects unanswered calls to a list of telephone numbers specified by the subscriber.

When unanswered calls are directed to Modular Messaging, the system checks to see if Find Me is enabled, and, if so, whether the call has arrived within an active schedule. If the call has arrived

within an active schedule, Modular Messaging asks callers if they want to leave a message or have the system try to locate the subscriber; in this case, the system prompts callers to record their name and then plays the Please Hold prompt.

¾ Caller priority level

◊ N/A


◊ N/A





◊ Find Me enables your mailbox to redirect unanswered calls to a list of telephone numbers. Your mailbox directs the calls in turn to each telephone number in the list, unless you answer. If you do not answer at any number in the list, the system asks the caller if the caller would like to leave a message.


◊ With Find Me, subscribers can set up schedules (times for rules) with an associated list of telephone numbers for forwarding unanswered calls. Find Me is implemented only for calls that are not answered because the extension rings, and there is no answer, and not for calls that are not answered because the extension is busy.

◊ When unanswered calls are directed to Modular Messaging, the system checks to see if Find Me is enabled, and, if so, whether the call has arrived within an active schedule. If the call has arrived within an active schedule, Modular Messaging asks callers if they want to leave a message or have the system try to locate the subscriber; in this case, the system prompts callers to record their name and then plays the Please Hold prompt.

One-X Speech Reach Me

Callers can initiate an outcall if the subscriber sets up the Reach Me number in one-X Speech. The Reach Me number is set up by the subscriber through the one-X Speech User Preferences Web pages. Subscriber authentication occurs before the subscriber logs on to the User Preferences Web pages.

The Reach Me feature allows a caller to locate a subscriber if the subscriber does not answer the initial call. The Reach Me interface allows a caller to reach the subscriber only at the Reach Me location defined by the subscriber. This prevents the caller from dialing other telephone numbers through the one-X Speech system.

If you use Microsoft Outlook, you can specify on the Reach-Me tab whether one-X Speech sends your e-mail notifications to your Reach-Me phone or your pager (or both). You

can also independently specify whether one-X Speech sends reminders of your Outlook tasks and appointments to your Reach-Me phone or your pager (or

both). Outlook users have these capabilities regardless of whether they have the Reach-Me feature for callers described in the bullet above. If you have

both Outlook and the Reach-Me feature, one-X Speech uses the same Reach-Me phone (or schedule of phones) for notifications and reminders as it uses for the Reach-Me feature.

¾ Caller priority level

◊ User can allow all callers or specific callers in a caller filter to attempt to reach them.





◊ User can Hold all calls, in other words, send all callers directly to voicemail

◊ You can say "What is my Reach-Me status?" to determine whether you are currently using or overriding the settings on the Reach-Me tab of your UCC SA User Preferences Web pages, and whether your calls are being forwarded to a specific number or to voicemail.


◊ User can specify a Reach-Me phone number that one-X Speech calls to try to reach you on behalf of your callers when your desk phone is busy or goes unanswered, or specify a schedule for using various such Reach-Me numbers.

◊ You can establish a Reach-Me phone number that is effective temporarily or indefinitely. This number overrides (disables) any Reach-Me number you previously specified on the Reach-Me tab of your UCC SA User Preferences Web pages.

If you defined and enabled a caller filter using the Reach-Me tab, that filter remains in effect throughout the "Follow me" override. Callers who are not in your filter list are sent directly to your business voicemail.

◊ Specify a personal operator to whom callers can choose to transfer


◊ User can create a schedule for when “Reach Me” is activated. The Reach Me feature allows a caller to locate a subscriber if the subscriber does not answer the initial call. The Reach Me interface allows a caller to reach the subscriber only at the Reach Me location defined by the subscriber. This prevents the caller from dialing other telephone numbers through the one-X Speech system.

There is additional information in 8.0.5.2 below where one-X Speech is discussed in detail.

8.0.5.2 Voice Portal Station uses should be able to use their telephone instrument to perform the following functions via voice commands:

• Set presence status and preferred phone

• Access unified messaging system for voice and email messages

• Access MS Outlook for contact lists, calendar, tasks, and other personal folders

• Initiate or join scheduled and ad-hoc voice or web conference calls with predefined workgroups


Describe voice portal features and functions available with the proposed UC solution, specifically addressing each of the listed voice command capabilities. Also identify other voice command operations supported by UC solution.




Avaya Response:

Avaya One-X Speech

Avaya’s one-X Speech lets mobile, remote and office workers easily access important communications tools and information via any telephone using simple and intuitive speech commands. one-X Speech customers can use any telephone and via speech commands gain access to calling and conferencing capabilities; voice, fax and email messages; enterprise directories and databases for contacts; and productivity tools such as calendars, contacts and tasks lists.

They can set personal contact rules to make more efficient use of their time - all via speech commands from any telephone.

Presence and Phone

◊ Setting presence status does not apply to one-X Speech, but you can identify where to be reached. User has the capability to:

o Specify a Reach-Me phone number that UCC SA calls to try to reach you on behalf of your callers when your desk phone is busy or goes unanswered, or specify a schedule for using various such Reach-Me numbers.

o Allow all callers or specific callers in a caller filter to attempt to reach you.

o Hold all calls, in other words, send all callers directly to your voicemail.

o Specify a personal operator to whom callers can choose to transfer.

In addition, if you use Microsoft Outlook, you can specify on the Reach-Me tab whether UCC SA sends your e-mail notifications to your Reach-Me phone or your pager (or both). You can also independently specify whether UCC SA sends reminders of your Outlook tasks and appointments to your Reach-Me phone or your pager (or both). Outlook users have these capabilities regardless of whether they have the Reach-Me feature for callers described in the bullet above. If you have both Outlook and the Reach-Me feature, UCC SA uses the same Reach-Me phone (or schedule of phones) for notifications and reminders as it uses for the Reach-Me feature.

Message Management

◊ Speech access to voice messages from Avaya Messaging systems, and email messages from Microsoft Exchange

◊ Ability to address messages using the voice mail directory, Microsoft Exchange contacts, contacts stored within the UCC Speech Access web application, or your corporate LDAP directory

◊ Play voice mails, send, forward, reply, and delete messages

◊ Read emails (via text to speech), reply, reply all, send, forward, delete, and save to a predefined folder

◊ Message navigation by message status (urgent, unread, read), media, and sender




Personal Efficiency Management

Management of personal contacts, calendar and tasks including Reach Me capabilities with ability to screen incoming calls, and set reminders while mobile.

Calling and Conference Management

Facilitates placing voice calls from any phone (via the corporate voice network) by speaking digits, using contacts from Exchange, from the UCC web contacts (Lotus Notes or Exchange), or corporate LDAP directory. Returning calls is easy with the ability to call the Sender of a call answer voice mail if the ANI is supplied or if they are part of corporate LDAP directory or personal contacts. It can even Call the Sender of an email if they are part of the corporate LDAP directory or personal contacts. It can even enable the creation of ad hoc conference calls while maintaining access to voice mail, email, calendar and tasks.

8.0.5.3 Text-to-Speech The proposed UC solution should support a text-to-speech feature that converts text messages to speech format for access from various voice terminal devices such as desktop telephones and mobile handsets.


Confirm the proposed UC solution supports text-to-speech and briefly describe the communications operations it supports.

Avaya Response:

Comply. Text-to Speech is supported in the one-X Speech application and Modular Messaging voice mail system.

one-X Speech

Avaya’s one-X Speech application provides the ability to use simple speech commands to greatly expand the power of communication and messaging capabilities. Great for when traveling or working remotely, the Avaya one-X Speech solution offers safe and easy hands-free access to important tools and information. While away from your office, use any phone to listen to voice mail messages, have your email read to you via text-to-speech conversion, and then reply and forward messages as needed. Make important phone calls initiate and manage conference calls, check and schedule appointments, set tasks and reminders—all using natural speech commands. One-X speech can also be used for automatic notification of important calls, messages and appointments, providing additional productivity and responsiveness to mobile associates. This application extends the same robust desktop capabilities you use in your office to wherever work takes you.




one-X Speech Text-to-Speech

The speech server software includes services and processes that run simultaneously and perform functions necessary to support the speech access application. Major services and processes of the one-X Speech platform consist of the following:

¾ Telephony engine

¾ Text-to Speech system (TTS)

¾ Automatic Speech Recognition system (ASR)

¾ Telephony server

The telephony engine hosts the virtual machine that executes the one-X Speech application. Based on instructions given by the one-X Speech application, the telephony engine accesses the following processes:

¾ TTS interface to translate text into speech that is played to the subscribers

¾ ASR interface to translate a subscriber speech into commands

¾ Telephony Server interface to answer calls, make calls, receive user instructions, and play prompts to the subscribers

Using Windows scripting engines loaded by the telephony engine, the one-X Speech application accesses COM/DCOM objects, database access methods, and run-time librariesto communicate with external systems, such as the voice messaging and e-mail servers.

A telephony engine process can support only one subscriber at a time. However, multipleengines can run simultaneously on a speech server. At large sites, dozens of engines can run at the same time—one engine for each incoming telephony port.

The TTS server receives text strings from other platform components. It applies dictionaryrules to the text strings, translates them into speech (audio stream), and returns them tothe requesting component.

The speech translation service is isolated in a separate component to improve performance and to allow for vendor independence for TTS.

The ASR system receives an audio stream from other platform components. It then loads an application or user specific grammar or both to the audio stream; attempts to recognize theaudio stream as speech, and returns the recognized text.

The telephony server manages the telephony subsystem. It routes incoming calls to an available telephony engine and provides:

¾ Calling

¾ Conferencing

¾ Transfer support

The telephony server, like the TTS and the ASR systems, uses a vendor-independent API. This API allows for independent protocols used by the telephony server.


on



Modular Messaging with Exchange Message Store (not proposed)

Text-to-speech is supported in the Modular Messaging voice mail platform enabling users to read text email messages via the Telephone User Interface (TUI). Text-to-speech (TTS) is inherent to the Modular Messaging system, providing for the conversion of text into speech (speech synthesis). Modular Messaging provides licensed speech synthesis software so that subscribers can hear the envelope and subject information of messages and text names over the telephone using a computer-generated spoken voice. Modular Messaging uses email readers for TTS conversion. TTS is also used for name confirmation when a recorded name is not available.

8.0.6 Mobile Clients Vendor Response Requirement:

What mobile clients are supported (Windows Mobile, Blackberry, et al)? What functions and operations can be supported on these mobile clients?

Avaya Response:

Avaya one-X™ Mobile

Avaya one-X™ Mobile builds on the functionality of Extension to Cellular by providing a family of mobile clients designed to further enhance the productivity of enterprise mobile workers. The family of Avaya one-X Mobile clients provides quick access to office phone functionality via an intuitive graphical user interface. The clients support devices ranging from high end smart phones to lower end feature phones to support most enterprise environments, independent of the mobile devices deployed or wireless networks used.

The Avaya one-X™ Mobile family consists of the following clients:

¾ Avaya one-X™ Mobile for Symbian

¾ Avaya one-X™ Mobile for Windows Mobile

¾ Avaya one-X™ Mobile for RIM

¾ Avaya one-X™ Mobile for Palm

¾ Avaya one-X™ Mobile for Java

¾ Avaya one-X™ Mobile for iPhone**

** - Available in US only in Q1 2008

Avaya one-X Mobile can be deployed in either “telephony only" integration or as part of a Unified Communications deployment.

Avaya one-X Mobile deployed in ”telephony only" integration provides single number access for both incoming and outgoing calls, as well as direct access to sophisticated features found

your office phone from your mobile device. The implementation is software based and only requires the purchase of Extension to Cellular software licenses added to

your existing Avaya Communication Manager server. The download and installation of the appropriate client is provided from the Avaya Customer Support website at

support.avaya.com. This deployment provides a robust integration with Avaya Communication Manager without

the need for additional hardware components.




Integration with Unified Communications applications such as voicemail, corporate directory, as well as access to additional sophisticated call handling and routing options may be required for some roles within the enterprise. To meet the need, This deployment includes the required software to integrate with enterprise applications and deliver additional capabilities to supported mobile devices through the required server hardware.

Avaya one-X Mobile features include:

One Number - Calls initiated by the mobile device will dial-by PBX and display the caller ID of the desk phone. Calls made to the desk phone extension will ring both the desk phone and the mobile device. The user can easily enable or disable both of these features from the mobile device.

One Voicemail - Unanswered calls will be picked up by the Enterprise voicemail system, eliminating the need to check multiple voicemail systems throughout the day.

One Device - The user can control whether calls are dialed via-PBX, for work-related calls, or dialed direct from the cellular device for personal calls, so users will not need multiple devices. Calls which are dialed via-the-PBX are made as though from the user’s desk phone. International calls will enjoy significant savings. Existing infrastructure, such as call recording, will remain useable even for mobile calls. Call detail recording can be used to distinguish mobile calls from desk calls for auditing of cell phone bills.

Handoff Calls Between Mobile Device and Deskphone - All calls made via, or received via, Avaya one-X Mobile can be seamlessly transferred between the desk phone and mobile device. An employee arriving at his or her desk while on a mobile call can pick up their desk phone to continue the conversation. While on a desk phone call, the user can become mobile by extending the call to their mobile device. In both cases, the handoff is seamless and is completely transparent to the other party.

Visual Voicemail*-Users can see corporate voicemail visually so they can prioritize messages and get to the critical ones first. Stored locally, there is no dialing into voicemail, saving minutes and time and the local actions are synchronized with the messaging server.

Sophisticated Call Routing* - Users can route incoming office calls to any device. Calls can be automatically routed based on schedule, location or on an ad hoc basis through a quick entry feature.

VIP Lists* - Mobile workers use VIP lists to manage how incoming calls are treated. Any calls not on the VIP Lists can be sent directly to voice mail to minimize interruptions.

Corporate Directory Integration* - Integration allows for easy access to corporate contacts through the search function and incoming callers id found in the directory are automatically translated showing the contact name instead of the number.

Advanced PBX Feature Access** - While on a call, the user has access to features such as conference, transfer, call park, and many more that IT selects. These are all accessed via a simple GUI.

Savings on International Calls - Direct dialed international calls from a mobile device can be very expensive. With Avaya one-X Mobile the user can leverage your existing corporate infrastructure from their mobile device. The user would dial the international number on their mobile device, but the international call would be dialed via the Avaya PBX. The mobile device is only making a local call to the PBX, with the PBX dialing the international number.




Business Continuity - With Avaya one-X Mobile, users only need to know each others normal desk phone extensions to remain connected. In the event that employees are unexpectedly unable to go to the office, everyone can remain connected and reachable.

Dual Mode Capability*** - Enables enterprises to leverage WiFi and Cellular networks with a single device. A user makes a call over cellular or WLAN. The call is seamlessly handed over as the user moves between networks. Both network connections provide access to all the normal one-X Mobile feature set. WLAN connectivity offers two main benefits:

¾ Cost Savings - When using a mobile device while on premise, call delivery path is WLAN so service provider minutes are not used.

¾ Increased Connectivity - Users can remain reachable at locations on premise without cellular service simply by installing WLAN access points.

*- Available only on RIM, Palm, Java & iPhone clients (iPhone available Q1 2008)

** - Available only on Symbian & Windows Mobile clients

*** - Available only with Symbian Dual Mode client

8.0.6.1 Mobile Security Vendor Response Requirement:

What security features are supported for mobile clients?

Avaya Response:

Avaya one-X™ Mobile provides the following security features:

Username and Password Policies

Avaya one-X Mobile for RIM, Palm, and Java integrates with Active Directory 2000 or 2003 for user authentication so a Single-Sign-On (SSO) can be deployed for the user name and password. This eliminates IT administrators having to administer and maintain another username and password for one-X Mobile. The Modular Messaging pin is used as another factor of authentication when users log into one-X Mobile.

Account Policies

Account Polices allow the Administrator to establish set system wide policies. These policies are applied to all one-X Mobile accounts. These Account Policies include:

¾ Disallow storage of voicemail on Mobile Device

¾ Forced Login

If voicemail storage is disallowed, then the dynamic download and playing of voicemails is allowed, however voicemails will not be stored on the mobile device.

If voicemail storage is allowed, then the user will have the ability to store up to 15 new and 15 saved voicemail messages on their device for playing. These files can not be forwarded from the application.

If the “Forced Login” account policy is set, the user will be required to enter their username and password each time they access the one-X Mobile application (Palm OS, and Java devices).




BlackBerry device users must enter their username and password if they have not used the device in the last 15 minutes.

When the “Forced Login” is not set, the username is cached and password cached/hashed on the mobile device after the user's first one-X Successful login. If the username and password are not able to authenticate on future login attempts, i.e. the information has changed since last login, the user will be prompted to enter the new username and password to gain access.

Login Failure

In addition to the system wide account policy the administrator can set a Class of Service account policy to require mobile users of one-X Mobile to login each time the application is launched. The one-X Mobile Application monitors failed login attempts, so if the user fails to enter the proper credential set three times, then one-X Mobile will obliterate all the mobile data, with the exception of the server URL and protocol setting. The user is also now locked out of the system for a preset time. The lock out increases in duration with each failed login attempt to ensure that any attack on the user’s account is rendered ineffective. In this state, one-X Mobile will continue to prompt for a login. If the user’s one-X Speech successfully logs in, one-X Mobile will automatically restore all the data that was obliterated.

Application Lock

The mobile client has an application lock feature. This feature allows a user to “lock” the one-X Mobile application so they can lend their phone to another party without fear that the other party will have access to their data or corporate voicemail. To unlock the application for use again, the user will need to reenter their user name and password.

Lost Mobile Device

The one-X Mobile Server does not associate a specific mobile device or set of devices with a particular account. Any mobile device can connect to any account if they use the proper credential set (username and password). Since there is no device specific device associated with an account, the account password is used as a special tag to help accomplish data obliteration.

If a device is lost, the user notifies the Administrator or goes to the End User Web to report the lost device. Both of these interfaces operate as follows:

¾ A warning dialog informs the administrator or user that they must take two actions to protect data on the lost mobile device. First the user needs to change their corporate password. Second they need to ensure that other mobile communication, such as corporate email, is disabled for the lost device.

¾ The data on the lost device is protected because the corporate password has been changed. When the one-X Mobile security module detects failed authorization attempts the one-X Mobile user data on the lost device will be obliterated.

¾ If the administrator has set the Forced Login security policy, then the mobile data will be obliterated when the login attempts fail. If the Forced Login security policy is not set, then the one-X Mobile client will automatically attempt to authenticate. When this authentication fails, the mobile data will be obliterated.




Additionally, on RIM Devices the administrator can use the “Erase Data and Disable Handheld” feature from the BES administration menu. In addition to wiping data used in conjunction with the BES server this will delete all the one-X Mobile data from the device and make the one-X Mobile application useless to anyone that recovers the device.

Account Deactivation

The Administrator is able to deactivate an account. After an account is deactivated, no access to the account is allowed.

Configuration / HTTP(S) Protocol Use

The one-X Mobile server can be implemented using four different topologies. All four topologies allow the administrator to mandate the use of a secure protocol (HTTPS) or allow an unencrypted protocol (HTTP). Mandating the secure protocol is done by deploying SSL on the one-X Mobile server(s). If you do not deploy SSL then HTTP protocol is used. Additionally for enterprises that use the BlackBerry client, the RIM administrator can mandate that all network traffic go over the BlackBerry standard encrypted channel to the BES (MDS). This ensures that BlackBerry security policies apply to all one-X Mobile communication.

One-X Mobile can be deployed as a single server inside the network behind the firewall or single server inside the DMZ. All incoming traffic from either the Mobile device or the web client is done on a secure port, either 80 or 443 (SSL) to the one-X Mobile server. If another layer of security is desired then it is recommend that either a reverse proxy server is used in the DMZ with a one-X Mobile server behind the firewall or a split server configuration is used with a one-X Mobile server in the DMZ and a one-X Mobile server behind the firewall.

A reverse proxy infrastructure hides the location of the Avaya one-X Mobile Server and the internal network structure from exposure to the Internet. Benefits of a reverse proxy configuration include:

¾ Only the external IP address of the Reverse Proxy is exposed to the Internet. Note that the DMZ address is still hidden.

¾ Using statically NAT’d addresses make it much more difficult for hackers to gather real information about the network.

¾ Reverse proxy infrastructure narrows the HTTP/HTTPS traffic origination to one server; the reverse proxy. Traffic can be monitored from one point.

¾ In order to hack into the internal network, a hacker will have to break through the following checkpoints:

◊ outside firewall interface

◊ Reverse Proxy

◊ DMZ firewall interface

◊ Avaya one-X Mobile Server

¾ If the Reverse Proxy is compromised, the hacker can only gain access to servers on the DMZ.




A split server configuration allows the Avaya one-X Mobile Server functionality to be split across two servers to keep only the parts necessary in the DMZ. The Avaya one-X Mobile Server Handset Component can be homed in the DMZ with port 80 or 443 (SSL) access from the outside. The Avaya one-X Mobile Server is homed on the inside with LAN access to the Avaya Communication Manager and other servers it needs to communicate with. Communication between the DMZ and the inside is limited to Port 8080 (Tomcat) and port 1433 (ODBC). Benefits of a split server configuration include:

¾ Incoming traffic from handsets can be easily load balanced in the DMZ.

¾ Access to port 80 and 443 is limited to the DMZ only.

¾ Firewall modifications to the inside interface is limited to 8080 and 1433. Critical Call Control functionality still resides on the inside next to the Avaya Communication Manager. This is important in load situations.

¾ VPN software may be installed on the Mobile Handset Components.

¾ In order to break into the network, a hacker would have to cross the following checkpoints:

◊ outside firewall interface

◊ Avaya one-X Mobile handset component

◊ DMZ interface on port 8080 and port 1433

◊ Avaya one-X Mobile Server

8.0.6.2 Web Client Vendor Response Requirement:

Is a Web-based client supported? If so, briefly describe its features and capabilities and how it differs from the standard UC client.

Avaya Response:

Comply. Web-based clients are supported by one-X Portal and the Modular Messaging Web Client.

Avaya one-X™ Portal

Avaya one-X™ Portal is a web-based software solution that provides users with control of telephony, conferencing, messaging, mobility administration, and access to corporate and personal contact information. Avaya one-X Portal consolidates access to all the critical voice applications that a user needs in a simple consistent interface. The solution complements the use of the desk phone while in the office, supports remote workers from any location, and is an ideal component of a business continuity strategy. It can be used by any employee from any internet connection if they cannot get to an office location in the event of a business interruption.

As a web-based solution, Avaya one-X Portal is implemented and managed centrally, and does not require software to be installed on user PCs. It brings together control of telephony (Avaya Communication Manager), conferencing (Avaya Meeting Exchange Enterprise), messaging (Avaya Modular Messaging with the Avaya Message Storage Server), allows the user to administer their Extension to Cellular settings and phone numbers, and access the corporate LDAP directory and personal contacts.




Users access one-X Portal by pointing their browser to the appropriate URL. While accessing the application from the internet, users interact with the system using a SSL VPN connection. Users can interact with one-X Portal in one of three modes:

¾ Shared Mode: The user initiates and manages calls from either their office telephone or the one-X Portal interface, while the office phone is used for audio. This is the ideal model for use while in the office.

¾ PC-Only Mode: The user initiates and manages calls from their PC, and the audio path is sent to the PC using voice over IP technology allowing the user to use the PCs multi-media capabilities to listen and speak. This mode is ideal when working in a hotel room, or any location where the user does not have access to a direct dial number to use for voice.

¾ Dual-Connect Mode: The user initiates and manages calls from the one-X Portal interface. To achieve toll quality voice, the user provides a telephone number where all calls should be directed. Inbound calls are directed to the designated number. When a user places an outbound call, the system first calls the user at the designated number, and then places the outbound portion of the call. This mode is ideal for a user working from a home or virtual office where there is an available telephone line with direct dial access in addition to the internet connection.

one-X Portal does not support call screening, web conferencing capabilities, or IP Video.

There is no one UC Client that supports all capabilities.

Modular Messaging Web Client

Modular Messaging with Message Storage Server (MSS), offers a Web Based Client that is for messaging only. When deploying Modular Messaging for Exchange, Unified Messaging

Systems; the native web interfaces are leveraged. When deploying Modular Messaging-MSS for voice mail and fax only, the Avaya Modular Messaging Web Client is utilized.

Modular Messaging Web Client provides:

¾ An integrated view of voice, fax, and text messages in the Modular Messaging mailboxes of subscribers.

¾ Message organizational capabilities, such as:

◊ Sorting messages by type, sender, subject, folder, or receipt time

◊ Searching for a particular message

◊ Moving messages within the inbox from one folder to another. Subscribers can move New messages to the Deleted folder or Saved folder, saved messages to the Deleted folder, and deleted messages to the Saved folder.

◊ Directory browsing

◊ Provides access to the Modular Messaging voice server directory so that from the directory, subscribers can search for other users and address messages to them.




◊ Toggle between telephone and local multimedia. Modular Messaging Web Client allows subscribers to toggle between recording messages by using local multimedia of their computer and using the telephone, regardless of the default settings

¾ Choice of language to view the interface

¾ Provides subscribers with a choice of languages to view the interface.

¾ Support Secure Sockets Layer (SSL) for secured access Modular Messaging Web Client supports SSL Web connections. However, customers must configure SSL on the Web server (as per the configuration supported by IIS) prior to installing the Web Client.

¾ Whether Web browsers connect by using SSL depends only on the Web server configuration and not on any settings in the Modular Messaging Web Client application.

¾ Access to e-mails on a corporate e-mail server

¾ Using the IMAP4 protocol, Modular Messaging Web Client provides subscribers with access to e-mail messages stored on a separate, corporate e-mail server. Subscribers can view, reply to, forward, create, send, save, and delete e-mail messages. Subscribers can also access the corporate LDAP directory to address email messages.

¾ Maximum message privacy enforcement

o When subscribers access messages that are marked private, Modular Messaging Web Client provides an appropriate indication to subscribers.

o Regardless of the system Privacy Enforcement Level setting for the VMD, Modular Messaging Web Client always enforces maximum message privacysubscribers cannot forward private messages.

o Administrators can configure Modular Messaging Web Client to support only dual-connect (telephone-playback only) mode. This prevents voice messages from being played locally on the computer, downloading voice messages, and caching voice messaging on the computer.

¾ Text subject creation - Subscribers using Modular Messaging Web Client can create a text subject when composing a new message. A text subject makes it easier for recipients to refer to the message when using a GUI client or scanning by means of the TUI. Subscribers can create a subject text when composing a message; subscribers cannot change subject texts once messages are sent.

¾ Edit message subject line - Modular Messaging Web Client allows subscribers to edit the subject lines of messages. However, a subscriber can edit the subject line only if the administrator has set the feature for the system.

¾ Address messages to PDL - Using Modular Messaging Web Client, subscribers can address messages to a PDL.

¾ Drag and drop messages - Subscribers using Modular Messaging Web Client can drag and drop messages into a Modular Messaging Web Client folder.

¾ Set message notifications - Subscribers can set Modular Messaging Web Client to notify them when they receive new messages.




The Modular Messaging Web Client inbox presents visual indicators, known as message flags that help subscribers to easily identify broadcast messages, priority messages, private messages, and delivery failures.

From Modular Messaging Web Client, subscribers can also send messages to the PDLs they own.

8.0.7 Microsoft Integration Vendor Response Requirement:

What is the level of integration with Microsoft Outlook, Office and Active Directory? Is the proposed UC solution compatible with the scheduled Microsoft Office Communications Server (OCS) solution? Describe any hardware/software requirements and options required to integrate with the Microsoft solutions.

Avaya Response:

Using Avaya Applications Enablement Services 4.0 (AES) which is a centralized server solution requiring nothing to be installed on the user’s PC or Avaya IP Softphone 6.0 which is a Windows client solution installed on a user’s computer Avaya Communication Manager can be integrated with Microsoft Live Communications Server 2005 (LCS) and Microsoft Office Communicator (MOC). Both AES and IP Softphone integrations provide integrated telephony and instant messaging presence, and click-to-call capabilities from Office Communicator and Microsoft Smart Tags. IP Softphone adds an Avaya tab to Office Communicator, providing direct access to advanced Avaya Communication Manager features. IP Softphone also adds additional capabilities to manage voice and video communications.

Avaya will be introducing one-X Communicator at VoiceCon, which is the next release of IP Softphone. At that time, Avaya will support Microsoft’s Office Communications Server (OCS) 2007.

Common Capabilities

Avaya Application Enablement Services or IP Softphone integrated with Microsoft Live Communication Server 2005 and Office Communicator provides the following capabilities:

¾ Integrated telephony and instant messaging presence. Presence status in desktop applications will show “Busy” or “On the phone” when an individual has initiated an outbound call, or accepted an inbound call. The presence is visible from the buddy list within Office Communicator. It is also visible from other Office Communicator windows such as the conversation window, and from applications that use Smart Tags such as Outlook or SharePoint.

¾ Click-to-call. As in the case with presence, the user has the option to “click-tocall” from the Office Communicator buddy list, conversation window, or applications that use Smart Tags such as the contact name in an email or SharePoint folder. Hence it is possible to read an email within Outlook, check the presence of the sender and recipients, click on a name, and initiate a call.

¾ Escalate from Instant Messaging to a Phone Call: Both solutions allow an exchange to begin as an IM session, and then have it become a live phone call.




¾ Call Control and Telephony Functions. Office Communicator provides an interface with feature buttons for managing calls. The AES integration supports the 17 telephony functions that Microsoft Live Communications Server supports with 3rd party PBXs via its TR/87 CSTA gateway. Application Enablement Services directly supports all of these capabilities within the Office Communicator Interface. IP Softphone supports all of the high priority capabilities with the Microsoft provided interface for Office Communicator, and the balance within Office Communicator by way of the Avaya Tab and other interoperability capabilities.

¾ Shared control of office phone: A call that is answered or initiated using either Office Communicator or the office desk phone can be managed with the other. That is, let’s assume that a user answers a call using the telephone. They can then place the call on hold using Office Communicator. They can then retrieve the call from hold and then forward the call using the buttons on the phone.

¾ Missed Call Notification: Both solutions will provide an email message indicating that a call has been missed. The message will include the calling line ID, date, and time of the missed call.

¾ Caller Popup Notifications: Both solutions will provide a popup “toast” window to alert the user to an inbound call.

IP Softphone Unique Capabilities

IP Softphone provides the following additional capabilities:

¾ Avaya Tab. Avaya has created many advanced telephony features driven by customer and market requirements. Many of these features can be used with hard or soft feature keys on the office desktop telephone. Avaya IP Softphone provides a tab within the Microsoft Office Communicator interface with direct access to these advanced telephony capabilities. The Avaya tab provides 12 feature buttons which can be associated with important functions to control calls. This includes the ability to manage simultaneous call appearances, and access to features such as Extension to Cellular, Record on Messaging, Transfer to Messaging, Consult, Whisper Paging, Call Park, Call Pickup, and many more.

¾ Expanded click-to-call: Click-to-call adds convenience for users because they can click on someone’s name and place a call. This enables the user to reach someone without needing to look up a number in a directory, and without interrupting the work flow. The user simply sees the name, and calls the person without hesitation. IP Softphone extends this capability to phone numbers in almost any Microsoft Windows application. In Internet Explorer, the IP Softphone software scans the pages and automatically detects phone numbers. It then highlights the number in yellow so the user is aware that they can click on the number. When the user moves the mouse over the highlighted area, the cursor changes to a phone indicating that they can “click-to-call”. In any other application a user need only highlight a phone number and then click the IP Softphone button in the system tray to launch the call.




¾ Mobility Support: While in the office, Office Communicator integrated with IP Softphone can share control of the office desk phone. When working at a remote location such as a hotel room, home office, virtual office, or customer location; IP Softphone can operate in one of two modes, leveraging the data network with or without the PSTN or cellular network to provide optimal voice quality. PC Only mode (a.k.a. Roadwarrior) enables the client software to have exclusive call and feature control while the voice is managed through the PC using voice over IP (VoIP). Any Phone mode (a.k.a. Telecommuter) enables the client software to have exclusive call and feature control while the voice is sent to a user-designated telephone. Inbound calls ring at the designated telephone; outbound calls first call the user at the designated number and then place the outbound portion of the call. All calls made or received maintain the business extension and name identity of the user regardless of their location or use of networks. Note that users can utilize Office Communicator integrated with Application Enablement Services, and use IP Softphone not integrated with Office Communicator or one-X Portal as the telephony end point to meet their remote needs.

¾ Call Logs: The time and date of all calls made and received while IP Softphone is running will be tracked in a log on the PC. The log will show the call type (inbound answered, inbound unanswered, outbound), name, number, and duration of the call. Users can enter notes in the log associated with specific entries. They can sort the log and extract the contents to be used with other applications such as Microsoft Excel or Microsoft Access for further management. Users can click-to-call from entries in the call log.

¾ Video: The Avaya Video Integrator streamlines video call setup using standard web cams. Video is managed through Avaya Communication Manager, so dial plan, features, and bandwidth management are coordinated together. The video software automatically identifies whether the far-end is video-enabled and establishes a video connection. Avaya IP Telephony features such as hold, transfer, forward and cover can be applied and synchronized with a video call. If desired, the video call can be dropped while the audio path remains active. Video connectivity can be established with any other user of IP Softphone with video, or any video user connected to a multipoint video bridge.




8.0.8 IBM Integration Vendor Response Requirement:

What is the level of integration with IBM Lotus Sametime, Lotus Quickr, and Lotus Web conferencing? Describe any hardware/software requirements and options required to integrate with the IBM solutions.

Avaya Response:

Avaya's Meeting Exchange solutions integrate with IBM's Sametime IM and Web conferencing. Sametime users can schedule sessions, notify participants via email, escalate IM sessions to conference, manage web and audio sessions in real time from a unified interface. Avaya provides a software adapter that is loaded on the Sametime server.

8.1 Conferencing & Collaboration 8.1.1 Ad Hoc Conference Vendor Response Requirement:

Is ad hoc and meet me audio/video conferencing supported? Briefly describe how such a conference is established and managed.

Avaya Response:

Avaya Meeting Exchange solution does support ad hoc and meet me audio and web conferencing, with video embedded within the web conference session.

Such sessions can be established by customer administration staff, and/or by end users. Avaya’s Client Registration Server (CRS) embodies a SQL 2005 relational database, and customer staff access the scheduling component by means of a Windows GUI.

In addition, Reservationless Account Profiles can be established and managed through standard LDAP integration into customer’s Active Directory. The account details of the reservationless accounts will be automatically emailed to end users. As employees are onboarded/off-boarded, end user conferencing profiles and privileges can be added or disabled automatically.

8.1.2 Scheduled Conference Vendor Response Requirement:

Can an individual station user schedule an audio/video conferencing session and reserve conferencing services as needed? Briefly describe how such a conference is established and managed.

Avaya Response:

Avaya Meeting Exchange solution does support individual users’ ability to schedule audio and web conferencing, with video embedded within the web conference session.

End users access the same scheduling component (CRS SQL 2005 mentioned above) via a browser based Web Portal (Apache Tomcat), which supports real time moderator conference control as well as scheduling.

Integration with IBM Lotus Notes/Sametime, Microsoft Office Communicator, and Microsoft Outlook are available for scheduling conference resources as well.




8.1.3 Host Control Features An audio conference host should access and implement, at minimum, the following features/functions: • View conference participants; • Play roll call of participants. • Mute all lines; • Mute select lines; • Disconnect select participants; • Dial out to bring new participants into the call; • Record conference; • Lock conference; • Initiate a sub-conference.


Briefly describe available audio conference host control capabilities, and confirm support of each of the above listed features/function (indicating implementation by TUI or Web-based control)

Avaya Response:

The Meeting Exchange solution supports all above feature/functions as host control capabilities. These features can either be done via the TUI or the Web-based interface (Web Portal).

The following host features are web-based:

¾ View all participants

¾ Mute all lines;

¾ Mute select lines;

¾ Disconnect select participants;

¾ Dial out to bring new participants into the call;

¾ Record conference;

¾ Lock conference;

¾ Initiate a sub-conference.

The following host features are DTMF/TUI based:

¾ Play roll call of participants.

¾ Mute all lines

¾ Dial out to bring new participants into the call;

¾ Record conference;

¾ Lock conference;

¾ Initiate a sub-conference.

¾ Apply gain to conference

¾ Emergency Blast Dial

¾ Billing Code prompt




8.1.4 Web Collaboration Vendor Response Requirement:

What form of Web services collaboration is provided for conference calls? Briefly describe how collaboration services are established and managed, including a listing of host control features and functions as they differ from audio-only conference capabilities. Indicate if recording and playback of an integrated audio/Web conference is supported.

Avaya Response:

Such Web Collaboration sessions can be established by customer administration staff, and/or by end users. Avaya’s Client Registration Server (CRS) embodies a SQL 2005 relational database, and customer staff access the scheduling component by means of a Windows GUI. End users access the same scheduling component via a browser based Web Portal. Integration with IBM Lotus Notes/Sametime, Microsoft Office Communicator, and Microsoft Outlook are available for scheduling conference resources as well.

The Avaya Web Conferencing solution offers:

¾ Slideshow push and annotation (*.ppt, *.jpeg, *.gif)

¾ White board design

¾ Chat: private or public

¾ Q&A sessions and polling capability

¾ Application sharing with all users

¾ Single interface for audio / data control

¾ Moderator streaming video

¾ Moderator control delegation

¾ Secure session entry – conference reference number & PIN access number.

¾ Synchronized Recording and streamed playback of the combined audio and Web session

¾ Exceptional Security – client/server connections are SSL encrypted, and no customer information resides on customer owned Avaya Web Conferencing servers prior to, or at the termination of, a moderator controlled conference

Conference Recording and Playback

With the Avaya Web Conferencing recording option, the Conference Moderator can choose to record the Audio and Data portion of a Web conference, and the Avaya solution will combine these two formats into one single file for post-conference streaming.

Users could leverage post-conference recordings for a wide range of uses, from e-learning to corporate training to sales presentations.

Features of the Web Conferencing Solution with Synchronized Record and Playback include:

¾ Simultaneously record the audio and web portions of the conference including the slide show

¾ Recording of all application sharing, and white boarding for a real-time replay of the meeting

¾ Management of recordings on a web interface for convenient user access




¾ Ensure security of the recordings by permitting — or prohibiting — access to specific users

¾ Play back recordings via Windows Media Player

Architecture Features

¾ Complete client to server security (https Port 443 using SSL)

¾ No client application downloads required, all standard Web based

¾ Lightweight browser based, Java base

8.1.5 Application Sharing Vendor Response Requirement:

Is application sharing services supported by the UC solution? If yes list the specific applications.

Avaya Response:

Yes. Avaya Web Conferencing supports application share mode. And Desktop share is supported as well, enabling virtually any desktop event of exploration (web browsing, e.g.) to be shared with session participants

- Documents

- Web Pages

- Remote PC control (subject to permission)

8.2 UC Architecture Vendor Response Requirement:

Briefly describe the hardware/software architecture of the proposed UC solution, including all necessary server and client software requirements.

Address in your response the following:

• Embedded open industry standards ands specifications, such as SIP, SOA Web services, et al.

• Level of design redundancy for both hardware and software elements, network interfaces, and memory storage

• Subscriber system capacity and scalability parameters

• Capability to support multiple PBXs (IPTS) and remote sites

Avaya Response:

One-X Speech:

UCC SA consists of platform components, software applications, and browser-based utilities.

¾ Standards

o Standard communication protocols, such as Internet Message Access Protocol 4 (IMAP4) and Simple Mail Transfer Protocol (SMTP), which provide interoperability with other systems

o Industry standard Microsoft Management Console (MMC)-based and browserbased utilities for administration




¾ Redundancy

There are two types of UCC SA configurations:

¾ Standalone configuration

¾ Multi-server configuration

¾ Avaya Messaging S3500 Messaging Server.

o This server supports both single sever (standalone) and cluster (multi-server) configurations.

o The current S3500 Server is a 3.4 GHz Pentium 4 Windows 2000/2003 server that comes with a NMS AG4040 series 2T/1600 telephony adaptor (46-port) , 2 GB of memory, 80 GB hard drive, 56K USB modem, and IIS Web server software. The customer must supply CDO installed from the Microsoft Exchange 2000 or 2003 System Management Tools (if implementation is to support Exchange) and pcAnywhere version 10.5+.

¾ Capacity

o The S3500 supports up to 24 Speech Access ports per server and up to 96 Speech Access ports when used in a cluster configuration. This server is only available in the U.S. and Canada. Or, customer may provide hardware, based on Avaya’s recommendations.

o The maximum number of administered subscribers on a single customer provided SA server is 5,000. Actual supported number of subscribers depends on the usage rate.

o Up to 48 Speech Access Ports may be licensed on a single customer provided SA server. A four-node cluster (4 cluster telephony nodes plus the cluster controller) can support up to 10,000 users and a total of 192 SA ports.

Note: this application’s capacity is primarily a function of speech resource limitations rather than processor limitations.

¾ Capability to support multiple PBX’s

o Can work with up to fifty different combinations of Avaya Modular Messaging, Octel 250/350, Octel 200/300, and Intuity Audix voice messaging servers, with or without Exchange or Lotus Domino e-mail support.

one-X Mobile:

¾ Standards

o One-X Mobile software runs on S8500C, with Microsoft Windows 2003 Server Service Pack 1 or high; Windows components of IIS and ASP.NET; Permission settings on IIS; time synchronization mechanism between the Avaya one-X Mobile server and the voicemail platform.

¾ Redundancy

o Not available.

¾ Capacity

o One AES and one one-X Mobile server supports 1000 clients; however, this number will increase over the next couple of months.




o Can support up to eight Avaya switches.

o No limit on number of Modular Messaging systems.

¾ Capability to support multiple PBXs

o Can support up to eight Avaya switches.

o No limit on number of Modular Messaging systems.

one-X Portal:

¾ Standards

Avaya one-X Portal is a software only offer. one-X Portal also requires one AES (either customer provided or Avaya supported)

Desktop Requirements:

Windows Desktop Software Requirements:

◊ Microsoft Windows XP, SP2

◊ Microsoft Internet Explorer 6.0/7.0, or Mozilla Firefox(1) 1.5/2.0

◊ Microsoft Outlook(2)

◊ Microsoft SMS client(2,3)

Also support Apple Mac Desktop

1) Mozilla Firefox does not support VoIP mode, Voice Recording or any other “Advanced” Client feature

(2) Only required for Portal Desktop Extensions Outlook/Exchange integration

(3) Required if the Portal Desktop Extensions are to be pre-installed on the PC via an SMS “push.”

¾ Redundancy

o Not Available.

¾ Capacity

o Can support up to 1500 clients, depending on server provided., supports up to 1000 clients; however, this number will increase over the next couple of months.


o Can support up to eight Avaya switches and multiple Modular Messaging systems.




One-X Desktop (SIP Softphone)

One-X Desktop consists of platform components, software applications, and browser-based utilities.

¾ Standards

o one-X Desktop uses SIP for telephony features. It also uses SOAP for contact synchronization with the PPM server on SES.

¾ Redundancy

o Avaya SES supports a duplicated server capability for high availability and advertises a shared virtual IP address to the rest of the network. During normal operation, all provisioned data and soft state such as SIP registrations and subscriptions are replicated between the primary and backup servers in real-time. Database replication and synchronization between the primary and back-up servers is accomplished via a dedicated Ethernet connection and direct cabling between the RS232 ports on each server.

¾ Capacity

o Users per S8500-based Combo/Home Server

o Up to 3,5000 users with 1GB RAM

o Up to 6,000 users with 3GB RAM

o SIP Stations per Avaya Communication Manager 5.0 S8300C with embedded Avaya SIP Enablement Services Combo/Home server

� Up to 100 SIP stations

¾ Capability to support multiple PBX’s

o Officially SES, but one-X Desktop has been successfully tested with Ubiquity, Asterisk & Cisco.

o Avaya Communication Manager can connect to a maximum of 16 Avaya SIP Enablement Services servers.

IP Softphone

¾ Capacity

o The maximum number of IP SoftPhones is dependent upon the Model of the Communication Manager and the server it resides on:

� S8700= 12,000, S8500=2,400, S8400=900 and S8300=75.

Meeting Exchange / Avaya Web Conferencing:

¾ Standards

◊ The Meeting Exchange conferencing system conforms to industry standards such as SIP (RFC3261 RTP RFC3550 DTMF RFC2833), G711, G723, G726, G729 codec’s, SOA Web Services, HTML; and with Avaya Communication Manager – analog/digital TDM, T1/E1, H.323; and with Avaya AudioCodes Gateways – T1/E1/T3




¾ Redundancy

◊ The Meeting Exchange conferencing components embody redundant power, fans, RAID1, dual NIC ports. In addition, N+1 server arrangements and extensive DR site redundancy configurations are supported.

¾ Capacity

◊ The capacity for a single 1U (1.75” H) 19” rack mountable audio conferencing server is 2000 ports, and such servers can be clustered with a media server to support up to 8000 ports in a single system. In addition, such systems can be deployed in multiple configurations. A single 1U (1.75” H) 19” rack mountable web conferencing server can support up to 750 web conferencing licenses. Server clusters can be configured to support up to 32,000 web conferencing users.


◊ The Meeting Exchange conferencing system is designed to support PBXs (and carriers) worldwide, with its standard SIP, non-proprietary protocol, as well as T1/E1/T3 with the addition of Avaya’s Audiocodes Gateways if needed.

◊ Avaya Communication Manager, for example, will transport analog/digital endpoints / T1, E1 Trunks / H.323 devices to the Meeting Exchange SIP platform, converting all disparate protocols to SIP.

23© 2005 Avaya Inc. All rights reserved.

Typical Meeting Exchange Enterprise Deployment

Communications Manager

Server and Gateway VoIP or Traditional Telephones

PSTN

Remote Users

Internet

LAN Switch

Firewall

Avaya Meeting Exchange

Avaya Web Conference Server

Active Directory Server CRS/Web Portal

SIP Trunks RTP streams

Video Telephony

¾ Standards

◊ The Avaya IP Softphone with Video Capability is a software architecture that is totally standards based, utilizing H.323, supporting H.261, H.263, H.263+




and H.264 video as well as G.711, G.728, G.729a, G.722, G.722.1,and Siren 14 audio.

Integrating with the Avaya Communications Manager requires a minimum version of 3.1.4 in the Communications Manager, and either the IP Softphone R5.2 with the Avaya Integrator for Polycom Video R2.0 or the IP Softphone R6.0, which includes the video integrator.

This solution give you a Unified dial plan for both voice and video, common signaling, common directories and common authentication and authorization procedures as part of the integration to the Communications Manager, as well as Bandwidth Management to prevent saturation of the LAN/WAN with privilege based prioritization allowing different bandwidth ranges for specific users or conference rooms.

The only hardware required is a standard USB Camera. IP Softphone also supports the IP hard phone and the Digital hard phone behind the Communications Manager.

The only requirement to activate the video on an existing or new Softphone is to open a port on the Optional Features, IP Port Capacities page, Maximum Video Capable IP Softphones, on the system-parameters customer options page 2 in the CM. Material Code 700365851 at a cost of $60ea.

The maximum number of IP Softphones with Video and Maximum Video Capable H.323 Stations combined is dependent upon the Model of the CM:

¾ S8700= 12,000, S8500=2,400, S8400=900 and S8300=75.

8.3 Systems Management UC systems management functions should include, at minimum, the following:

• Station user telephone instrument management

• Status displays for all UC system components

• Audit trails (configuration log)

• System resource monitoring (CPU, memory, disk space)

• Serviceability, error logging, and tracing features

• Storage & Reporting features

• Interface for backup and recovery


Briefly describe UC systems management tools, including hardware/software requirements, and confirm that support for each of the above operations.




Avaya Response:

One-X Speech

Administration of the UCC SA Web applications, such as Speech Server Reports, requires Microsoft Internet Explorer 6.0. The administrator cannot use Netscape to administer the UCC SA Web applications.

UCC Speech Access User Preferences is a browser-based utility that allows subscribers to change their application settings and access user documentation.

UCC Speech Access User Management (SAUM) is a browser -based utility that allows you to perform user administration functions, such as adding or removing subscribers and updating their records, and setting up newsletter delivery (for Microsoft Exchange subscribers only).

UCC Speech Server Reports (SSReports) is a browser-based utility that uses Active Server Page (ASP) to view reports about user sessions. You can use these reports to monitor system activities and make tuning decisions. Also have the capability to monitor the Speech Server or Server Set to ensure that it is running and communicating with other systems, as well as monitoring platform disk space.

UCC SA offers an outbound call detail report through the UCC Speech Server Reports, a browser-based application that queries the UCC SA database and generates reports, ranging from system performance to user-related statistics

The outbound call detail report provides information on all outbound calls that are placed by a subscriber during a specified period of time.

The administrator can run the outbound call detail report to monitor toll fraud or outbound call usage by specific subscribers. The report also alerts the administrator of unusual toll usage.

There are also log files to assist in troubleshooting issues.

one-X Mobile

The Serviceability section of the Avaya one-X Mobile Administrative Interface allows monitoring and management of services, and provides information about Avaya one-X Mobile Server components and allows fine tuning of these components.

Trace Components allow you to manipulate the data stored in logs. The default setting on each field on this page is to debug. This provides the most information and will also allow you to troubleshoot more thoroughly.

Component Info provides information about various components such as version numbers and last known status.

With the Avaya hardware, we include Ghost which allows a backup of the server

one-X Portal

The one-X Portal Administration application contains the Administration Command Line Client and Administration Web Client application, These applications enable you to administer the users and services and system tasks on one-X Portal.

The Administration application is used by Administrative users and Auditor users. The security groups for these users are added and configured for one-X Portal in the installation and implementation process. The users themselves can be added or deleted at any time.




Administrative users

Users who administer the users, servers, and system functions on one-X Portal. Administrative users have permissions to perform all of these administrative tasks using the Administration application.

Auditor users

Users who have read-only privileges and restricted access to the functions in the Administration application. These users are typically tasked with reviewing changes to one-X Portal but do not have the authority make changes themselves. If an Auditor user tries to change a record, the Administration application returns an error. The Scheduler and Monitor functions restrict access while some of the other functions return an error after the Auditor tries to change a record.

The Administration Web Client supports the following one-X Portal administration:

¾ Management of Portal and Prototype Users

¾ Control of feature availability through configuration and scheduling

¾ Configuration of system components

¾ Provisioning of system resources

¾ Maintenance of system operations

¾ System status

System Administrator have tools for troubleshooting to help you resolve issues on one-X Portal. There are solutions for the one-X Portal client, the Desktop Extensions, the Administration application, and general one-X Portal issues.

Troubleshooting also provides descriptions of the SNMP Traps (event notifications) generated by one-X Portal by component. These components include Licensing, Scheduler, Database Backup, Conferencing services, Telephony services, User services, Enterprise Directory Synchronization, and Modular Messaging Synchronization to name a few.

Avaya one-X Portal generates alarms (SNMP traps) to notify users of system events. Alarms are grouped by categories. Each alarm category identifies the system component that generates the alarm. Most of the alarms are displayed in log message files.

one-X Portal also has backup capability.

Meeting Exchange/Avaya Web Conferencing:

The Meeting Exchange solution supports system management functions allowing administrators access to application with extensive reporting, logging, back-up and storage capabilities.

¾ SNMP is supported. Alarms are grouped by categories. Each alarm category identifies the system component that generates the alarm. Most of the alarms are displayed in log message files.

¾ Utility for on board back-up is available, as well as support for file transfer to customer network file server

¾ SSH secure access is supported for customer and Avaya remote staff

¾ Customer approved VPN arrangements are supported, as well as dial up modem access if customer requirements dictate. PPP dial up to secure LAN is also supported.




The Meeting Exchange solution supports system management functions allowing administrators access to application with extensive reporting, logging, and back and storage capabilities.

8.4 Security What security features and functions are embedded or are available with the proposed UC solution? In your response specifically address each of the following security issues:

• Password Access

• Authorization

• Authentication

• Encryption

• Certification strategy

Avaya Response:

one-X Mobile Security

Please refer to Section 8.0.6.1 for security details.

one-X Speech.

Subscriber access to one-X Speech is protected by password schemes. One-X Speech retrieves subscriber accounts from one-X Speech database by matching the subscriber one-X Speech account numbers with their voice mailbox IDs. One-X Speech uses the subscriber touch-tone (DTMF) or spoken password, together with the mailbox ID, for authentication on the voice messaging server.

One-X Speech locks the account if the subscriber repeatedly fails to provide the correct password. The administrator can configure the number of consecutive login failures that determines the lock-out of a subscriber account. The administrator can also configure the duration for which the subscriber account stays locked.

one-X Portal:

Avaya one-X Portal provides you with options for a secure implementation.

¾ Password Access.

o Secure connections through firewalls

o Each user has their own password.

¾ Authorization.

o User authentication through the enterprise directory

o Access to one-X Portal through secure server connections

¾ Authentication.

o Connections to integrated components through secure ports

¾ Encryption

o Encryption implemented through the Administration Command Line Client

¾ Certification strategy




o As new releases of Microsoft Exchange, IBM Lotus Domino, Avaya Modular Messaging, Intuity Audix LX, and IA770, testing is performed to ensure functionality and capability

one-X Desktop:

The Avaya one-X Desktop addresses security in the following ways:

¾ Password Access. Passwords are managed by the SES server, and users can change their own password via the web-based SES User Interface.

¾ Authorization. Only authorized (as in successfully logged in) one-X Desktop users can make calls. However, we do support a peer-to-peer mode, and that does not require authorization. However, you can't register to a proxy, nor can a SIP Softphone registered to a proxy receive a call that did not come from the server (as in a peer to peer call).

¾ Authentication. Authentication is done via the SES server. When registering with SES, the one-X Desktop authenticates with SES, which owns the credentials. In conformance to the SIP standards, MD5 is used to encrypt the password during login. See Password access above as well.

¾ Encryption. one-X Desktop authenticates with SES using the recommended algorithm with MD5. In addition, all signaling traffic is sent over a TLS link, which is encrypted.

¾ Certification strategy. If you're talking about interoperability, not only does our Solution Interop Lab test with other SIP vendors & providers. Many Application Notes are publicly accessible.

IP Softphone:

As an IP endpoint client of Communication Manager, all conversations can be encrypted and secured. IP Softphone works with a variety of VPN solutions, including Avaya VPN remote Client software and Avaya SG200/203/208 Security Gateways, fulfilling the promise of secure communications across the breadth of an enterprise network.

Security Features:

¾ Survivability against Denial of Service Attacks.

¾ Password protected login sessions.

IP Video

The Solution leverages the security of Avaya Communication Manager, including: Real Time Media Encryption, Access Security Gateway, Malicious Call Tracking, Toll Fraud protection, and Emergency Calling Services (i.e. E911). See Avaya Communication Manager product information for more security details. Polycom video endpoints have been COMSEC tested by Titan Systems, Information Security Systems Division, and independently validated to operate with approved government encryption technologies. See Polycom VSX and MGC fact sheets for more security details.

Meeting Exchange/Avaya Web Conferencing

¾ Password Access.

o Conferences are password protected by moderator and participant codes. Reservationless conference codes can be automatically changed on a selected duration interval. Administration access is also password protected.




¾ Authorization.

o Hosts can authorize participants to join their conference by giving out the credentials for the conference both for audio and web conferences.

¾ Authentication.

o Host can turn name/record/playback on to determine participants and also have a visual representation of who is on call using audio console within the Web portal product.

¾ Encryption

o Clients using web conferencing are encrypted through SSL. SIP call control is encrypted via TLS links.

¾ Certification Strategy

o Avaya builds applications within industry standard security guidelines. Support Engineers monitor third party sites for security/certification updates. As new releases of integrated third party software become available, testing is performed to ensure functionality and capability.



Part 1 - Section 9 - Automatic Call Distribution Contact Call Center

Section 9: Automatic Call Distribution (ACD) Contact Call Center 9.0 Basic ACD Call Center Requirements VoiceCon requires that the proposed IPTS communications solution must support an ACD-based contact call center solution that includes call screening, call prompts, automatic distribution routing, call queuing, announcements, call handling, agent mobility, management and reporting, feature configuration and programming, administration.


Confirm your proposed communications system can support the listed functional requirements and briefly describe the necessary hardware and software requirements. Indicate if any ACD capabilities are embedded in the IPTS generic software package.

Avaya Response:

Comply. Avaya’s proposed ACD solution will support an ACD-based contact call center solution that includes call screening, call prompts, automatic distribution routing, call queuing, announcements, call handling, agent mobility, management and reporting, feature configuration and programming, administration.

Avaya Call Center Software can run under the Avaya Communication Manager firmware running on an Avaya Server and Avaya Media Gateway configurations the hardware discussions are elsewhere within this document.

Because Avaya Call Center Basic is included Avaya Communication Manager 5.0 system without requiring a separate ACD system, server, hardware, or software then all the ACD capabilities provided by Avaya Call Center Basic would be automatically embedded in the IPTS generic software package at no additional charge.

Call Center Basic includes basic ACD functionality because it includes call queuing, basic announcements, direct (linear) call distribution or uniform call distribution (most idle agent

hunting), agent login/logout, agent work states (Auxiliary Work, After Call Work, Auto In, Manual In, Auto Available, Auto Answer), agents logged into multiple splits (think “hunt groups”), multiple call handling, Service Observing, VuStats, and more.

Avaya also includes some basic reporting in the form of Avaya Basic Call Management System (BCMS) which provides “builtin” Call Center reporting capabilities without requiring any adjunct server or software. Avaya Basic Call Management System provides a variety of measurements that may be used

to monitor the operations of an automatic call distribution (ACD) application. Each type of measurement data is processed and organized into a unique report. The primary purpose of these reports is to provide information useful for managing ACD facilities and personnel. In addition, Avaya Basic Call Management System is an inexpensive “back-up” for users who are taking advantage of the exceeding popular Avaya Call Management System (CMS).

In the interest of being concisely accurate to the requested VoiceCon requirements, Avaya is choosing to respond with an a la carte offered solution. However we do want VoiceCon to be aware of the fact that Avaya offers, on a worldwide basis, two (2) Customer Service Editions which are full-service company contact center solutions delivering proven software and components for intelligent routing, contact management, self-service applications, and operational performance. There are currently two editions available, Standard or Advanced. Each edition is designed to meet specific configuration and capability needs:




¾ Avaya Customer Service Standard Edition is a complete company voice call center that includes skills-based data-directed routing, remote agents, premier reporting, desktop integration, self service, and outbound preview dialing.

¾ Avaya Customer Service Advanced Edition enables execution of more sophisticated company routing and any-media customer service strategies. This edition includes all the capabilities of the Standard Edition plus SIP-based multimedia handling, patented predictive routing and resource selection algorithms, fully customizable desktops with support for any media, and predictive dialing.

Please see Avaya Appendix 5 Call Center Options for a listing of available offers.

9.0.1 Third Party System Integration VoiceCon requires that the proposed ACD solution be able to support third party equipment.


Confirm that the proposed ACD solution can support each of the following third party equipment options:

• Message Board

Avaya Response:

Comply. Avaya’s proposed ACD solution can support third-party message boards.

Through the Avaya Developer Connection Program partners Avaya can provide integrated solutions in this space.

• Interactive Voice Response (IVR) system

Avaya Response:

Comply. Avaya’s proposed ACD solution can support third-party IVR systems.

• Customer Relationship Management (CRM) system

Avaya Response:

Comply. Avaya’s proposed ACD solution can support third-party CRM systems.

• Workforce Management system

Avaya Response:

Comply. Avaya’s proposed ACD solution can support third-party Work Force Management systems with or without their optional Agent Adherence modules. Avaya, through our existing partnerships can resell some of the more popular systems such as IEX and Witness 360 in today’s marketplace to VoiceCon and will act as a single point of contact for those items.

• Quality Monitoring system

Avaya Response:

Comply Avaya offers products from several industry leaders and while we still offer at least one ACD telephone set (CallMaster) designed for call center agents that contains a built-in RIM (Recording Interface Module), the days of widespread use of RIMs in this industry are over.

Avaya’s proposed ACD solution can support third-party Quality Monitoring systems.




The term quality-monitor encompasses a wide variety of solutions but most quality assurance efforts are still very voice oriented. In this space, Avaya offers products from several industry leaders such as Nice, and Verint (Witness) in the area of “agent performance”. While Avaya still offers at least one ACD telephone set (CallMaster) designed for call center agents that contains a built-in RIM (Recording Interface Module), the days of widespread use of RIMs in this industry are over.

Avaya’s stated direction, based upon marketing research coupled with a review of the competitive actions of those who create telephone sets for agent use it is no longer vogue to include RIMs in agent telephones. Recording via LAN/WAN is preferred and helps centralize storage and retrieval of conversations held anywhere in the enterprise. Our customers are completely in synch with this shift Avaya has made over the last several years in this area.

Avaya can also offer solutions from our DevConnect programs such as Click-Fox in support of more sophisticated quality adherence tracking.

9.0.2 ACD Station Equipment VOICECON requires that the proposed ACD solution be capable of supporting a mix of terminal equipment for agents and supervisors.


Confirm that the proposed ACD solution can support a mix of ACD agent/supervisor station equipment that includes analog and IP desktop telephone instruments and PC client soft phones for agent/supervisor voice communications requirements.

Avaya Response:

Comply. Avaya’s proposed ACD solution is capable of supporting a mix of terminal equipment for agents and supervisors.

Avaya’s proposed ACD solution, comprised of an Avaya Communication Manager 5.0 with Avaya Call Center Software – Elite 5.0 can support a mix of ACD agent/supervisor station equipment that includes analog, DCP (digital) and IP desktop telephone instruments and PC client soft phones for agent/supervisor voice communications requirements.

In addition to analog, digital, and IP desktop telephones, Avaya also supports SIP desktop telephones for call center use as well. Existing and potential customers looking for a lowcost SIP agent desktop telephone called the 16CC may choose to avail themselves of this new SIP only set.

Please See Avaya Appendix 3 –Slide 34 for photo

9.0.2.1 ACD Telephone Instrument Vendor Response Requirement:

Briefly identify and describe all desktop telephone instruments and equipment in your portfolio that are designed specifically for ACD agents and supervisors.

Avaya Response:

Please take a moment out and review section 4.2 regarding Desktop IP Telephone Instruments. While Avaya realizes that VoiceCon has a requirement for several types of desktop IP telephone instruments for various user personas (such as Economy, Administrative, Professional, Executive, and Call Center) it is also true that in addition to the SIP ACD set called the 16CC, that Avaya supports any 2500 style analog telephone, DCP (Digital Communications Protocol) telephones or IP (H.323) telephones in use to support VoiceCon 08 staffed ACD agent and ACD supervisors.




Our recommended Call Center Set that would be used by both the VoiceCon ACD agents and ACD supervisors. The Avaya 9650 IP Telephone (please see Avaya Appendix 3, PowerPoint Illustrations for photos) is designed to meet the needs of almost any Professional or Executive user as well as your Call Center users (Agents and/or Supervisors) given it ability to support up to 24 call appearance/Communication Manager features and can be used with the optional SBM24 expansion module. This module provides 24 additional programmable buttons, which may be configured as call appearances, bridged appearances, or feature access keys. The SBM24 Expansion Module is powered from the telephone via the module interface port, so the expansion module does not, in and of itself, require a separate power supply.

The Avaya 9650 IP telephone provides 11 one touch buttons for quick access to Communication Manager Features or Call Appearances. The Avaya 9650 IP Telephone exceeds the requirements listed except for the inclusion of a Recording Interface Module (RIM) interface. RIM is mostly now done over IP via a Nice or Verint/Witness solution especially where Sarbanes-Oxley is concerned. If VoiceCon requires a traditional RIM at the desktop then other features may be lost such as the full-duplex speakerphone.

Since the recording need can be met by use of an optional Avaya supplied offer from one of our valued partners, we can say that 30 out of these 30 comparative items are met (100%) as opposed to only 29 out of 30 comparative items being met (96.6%) with the RIM not being supported in the Avaya 96xx telephone. VoiceCon would need to supply some sort of recording device per Agent per phone a Centralized IP solution would likely be recommended. In solution discovery, our client executive, or solutions architect would engage in the appropriate “application discovery,” taking into account VoiceCon’s business plans prior to recommending a “recording solution from Avaya.”

We do have a number of additional telephones that have designed specifically for ACD agents and supervisors but are not IP so they have not been proposed in this response. The Avaya CallMaster® V (DCP) has been specially designed to support applications involving the Automatic Call Distribution (ACD) feature of the Avaya Communication Manager. The ergonomic design of the CallMaster V enables agents to handle large volumes of calls more quickly, efficiently, and productively—in customer service, order processing, collections, account management, or any communications-intensive activity. VuStats’ display of agent and call center statistics on the CallMaster V provides agents with real-time information they can use to improve their own performance and that of the Voice Con call center.

The CallMaster V has the same look and feel of the standard Avaya 6400 Series telephones but there are several significant additions that maximize the value of this telephone in a Call Center environment:

¾ 2-Built-in Headset Jacks – The CallMaster V is designed to use Avaya headsets.

¾ Built-in Recorder Interface Module (RIM) with Warning Tone – Will support recording of both the agent’s and caller’s voice on a voice activated analog tape recorder. A soft beep warning tone is repeated every 13.5 seconds to notify the agent and calling party that the call is being recorded (user can deactivate).




The CallMaster V is designed to work on a 16- or 24-port, 2-wire Digital Line Circuit Card and since VoiceCon is requesting an IP solution this would add unneeded hardware to support the units.

Please see Avaya Appendix 3, PowerPoint Illustrations slide 34 for photos.

9.0.2.2 Supervisor Workstation Vendor Response Requirement:

Briefly describe the proposed supervisor workstation solution in your ACD solution proposal, including telephony and ACD-specific feature and functions, toolbars, and report screens.

Avaya Response:

Avaya Call Management System (CMS) Supervisor software for VoiceCon’s supervisor personal computers provides access to all standard and custom reports, graphical and textbased, available under CMS. And now, these reports are even more “business useful” with the following customization enhancements:

¾ Sorting capabilities that let supervisors rearrange report information in an order that is easy for them to use. Each supervisor can customize reports with up to three (3) sorting levels, such as alphabetically, by agent/state, numerically by time/agent, and more.

¾ More options for call center analysis with the capability to mix real-time and historical information together on the same standard report. The Avaya term for a report that contains both real-time and historical information in the same report is called an “integrated report.”

¾ New Graphical Supervisor reports are easier to interpret and more meaningful thanks to improved graphical display capabilities, including colors.

¾ Ability to customize reports by selecting data, formatting, positioning, and the type of charts (including pie charts, 3-D bar charts, line graphs, and simple grids).

¾ Customized threshold and exception alerting help call center managers rapidly respond to changes within the call center

¾ CMS Supervisor lets you create call center reports in the colors and format that work best for you, and import reports or data between CMS and other applications as needed

Avaya Call Management System (CMS) Supervisor real-time reports give VoiceCon supervisors snapshots of the call center’s performance and status. Abandoned calls, for example, can be monitored to determine the waiting-for-service tolerance of callers and compared to the number of calls in queue. Additionally, agent productivity can be compared at a glance to determine who may need help in speeding up their typical duration of aftercall work (ACW) time.

Over 40 real-time reports are available in a variety of easy-to-interpret graphical and textbased formats that can be displayed on your PC, printed, stored to a file, copied to a clipboard, run as a script, or exported to HTML format through the Save as HTML feature. Standard real-time reports display data for the current interval for agent, split/skill, trunk/trunk group, vector, and VDN activities, such as number of ACD calls, abandoned calls, average talk time, and so on.




You can use Avaya CMS’ reporting capability to get real-time information, updated as often as every three (3) seconds, depending on the user permission and number of active terminals and open windows that will help you monitor ongoing performance and status so you can make any necessary adjustments quickly.

Avaya CMS Skill Status Real Time Report

In the CMS Supervisor real-time skill status display shown below the chart format and sorting order of the staffed agents has been changed from the pie chart shown above and these agents are sorted in ascending time order. Threshold highlighting is also illustrated showing agents spending excess time in Auxiliary (AUX) Work state, a Service Level Warning, and an Oldest Call Waiting caution level. These are obviously very useful in the minute by minute management of a call taking center.

Avaya CMS Skill Status Real Time Report




Avaya CMS Agent Information Drill Down Report

Double clicking on an agent name in the report allows a supervisor to drill down and access individual agent information.

Avaya offers the ability to easily customize the CMS real-time and historical reports through its professional services organization known as Avaya Consulting and System Integration.

Avaya Operational Analyst

Avaya Operational Analyst is an optional full-featured multi-channel contact center real-time performance monitoring, historical reporting and sophisticated analytical system supporting Enterprise businesses that need integrated and consolidated operational data storage, reporting and analysis for the Contact Center with an Avaya Call Management System. It is not a good fit for Voice Con 08 as it is more typically deployed in an enterprise that has more then one CMS and wishes to consolidate reports from these in a central repository.

Avaya Operational Analyst is often deployed as part of an Avaya Interaction Center, functioning as its operational data store and contact center performance analysis system, provide reporting and real time monitoring.

The integrated customer interaction repository and suite of standard predefined reports provide detail and summary views across multiple sites, multiple ACD vendors and multiple communication channels (Voice, VoIP, email, Web Chat and Web Self-Service). The 3-D graphical, web-based monitoring of real time and historical contact center performance allows for efficient management of agent activities and verification that the system is achieving service levels goals. The ability to create and maintain customized reports aids in the assessment of customer service and marketing activities.

The Operational Analyst Real-time Event Processor collects and processes real time events from an Avaya Interaction Center. Data from agent desktops and the multi-media channels are collected and processed to provide comprehensive and consistent real-time reports across all channels. In addition, different views of real-time data are provided including the current 30-minute interval and up to four (4) user-defined 24-hour views.




Operational Analyst Report for

Agent Performance by Service Class and Queue

Operational Analyst Report for IC Agent Performance




9.0.3 Remote ACD Station Equipment VoiceCon requires that the proposed ACD solution can support remote agents.


Confirm that the proposed ACD solution can support remote off-premises IP desktop telephone instruments and/or PC client soft phones for agent communications requirements. Also indicate if your system solution can guarantee Quality of Service levels for PSTN level connections to remote agents.

Avaya Response:

Comply. Avaya’s proposed ACD solution, comprised of an Avaya Communication Manager 5.0 with Avaya Call Center Software – Elite 5.0, can support remote off-premises IP desktop telephone instruments and/or PC client soft phones for agent communications requirements.

Additionally when remote agents are supported by Avaya IP Agent in the “telecommuter mode,” rather than the “road warrior mode,” our proposed system solution has little need to guarantee Quality-of-Service (QoS) levels for PSTN level connections to these remote agents since the lines used for the voice portion of the call are already there (such as local residence or business lines obtained from a local exchange carrier and already at PSTN toll quality). We will recap more of this when we answer the subsequent question in this section as to “Can your system guarantee Quality of Service (QoS) at the PSTN level for the voice channel for remote agents, and if so how is this done?”

9.0.3.1 Virtual Contact Center VoiceCon may require a virtual contact call center solution in the future as it integrates CHQ facilities with other corporate facilities.


Confirm that your proposed ACD solution can support a virtual call center environment, and include in your response answers to the following questions:

Avaya Response:

Comply and since this RFP specifies the use of “skills based routing” Avaya’s proposed ACD solution is equipped with Call Center Software – Elite. Avaya Virtual Routing [Multi-Site Best Service Routing (BSR)] is included at no additional cost within the Avaya Call Center Elite Software as proposed.

When VoiceCon is ready you will just need to ensure that you can meet the network requirements to deploy Multi-Site Best Service Routing. Your networks must meet a number of criteria:

¾ To ensure your existing network will meet the requirements for BSR support it is recommended that you contact your Client Executive about having Avaya perform BSR network certification.

The network must support end-to-end transport of codeset 0 user data, either as a User-to-User Information Element (UUI IE) or by QSIG Manufacturer Specific Information (MSI IE), in the ISDN SETUP and DISCONNECT messages. The network must also allow User-to-User Information Elements (UUI-IEs) to be transported in the first call-clearing message prior to answer, while a call is still in the call-proceeding state. With BSR poll calls, the information is forwarded back in the DISCONNECT message. In this case the network must support forwarding of UUI in the first call clearing message prior to the active state (in other words, while still in the call proceeding state).




¾ Private networks can be configured for either QSIG (transport using MSI packaged in a codeset 0 Facility IE) or non-QSIG (transport using a codeset 0 UUI-IE). Currently, public networks do not support QSIG and user data can only be transported using the UUI-IE when supported by the network. Future public network offerings will undoubtedly support QSIG, possibly by Virtual Private Network (VPN).

¾ The communication server must support the ISDN country protocol.

¾ The network byte limit for user information contents (the user data portion) must be large enough to carry the data needed for the customer application.

It is important to note that some public network providers may require service activation and/or fees for user information transport. It is imperative that the response time for ISDN signaling should be fast enough that an entire consider series can execute within the span of a single ring cycle. Enhanced information forwarding has been tested with several major carriers. To find out if these capabilities work with your carrier, check with your account team for the most current information. If testing has not been done to verify operation over the public networks involved with the preferred specific configuration, use of private ISDN trunking between the nodes should be assumed until successful testing has been completed.

• How many remote sites can be supported?

Avaya Response:

The answer to the quantity of remote sites that can be supported off a single instance of Avaya Communication Manager based is actually based upon how VoiceCon defines a “remote site”. Each extension off the proposed Avaya Communication Manager from CHQ1 could be at a unique remote location

However the more traditional view of a remote site is a site specific location of connectivity that can handle both local trunk terminations and support for local extensions and call taking agents. So under this perspective the “more traditional Avaya answer” is that a maximum of 250 remote sites can be supported.

• How remote sites are physically and logically supported?

Avaya Response:

There is only one software instance of Avaya Communication Manager running Expert Agent Selection (EAS) at CHQ1 and each of the remote sites be supported by their own Avaya Media Gateway and some form of LSP each with their own trunks and extensions. An agent can be placed at any location, even a remote workers site (home office) up to the purchased licenses VoiceCon has requested.

As part of the added value of this solution, if VoiceCon opted to move their 115 active and 200 configurable agents who are supporting 15 agents skills out of the “main” call center location at CHQ1 and now have perhaps 50 agents working out of CHQ1 with another 50 at CHQ2, another 50 at RO) and yet another 50 at SO. These agents are all off the same Communication Manager instance (by design) and there is no need to implement Avaya Virtual Routing [Multi-Site Best Service Routing (BSR)] to load balance call routing and resource selection as all the 200 agents assigned to Skill #1 (Service) can be all equally considered for both call and agent selection. However if you want to place your Call Center trunks at all four (4) locations (CHQ1, CHQ2, RO and SO) you could opt to activate Location Preference Distribution which means that you can give preference to connecting the staffed EAS agent.




By this we mean that with Location Preference Distribution active, when a call arrives on a trunk located in the particular office the call can be routed to an agent in that office as the primary choice with the desired skill first. This can greatly reduce calls transversing your network.

• Can call loads be balanced across multiple sites to avoid agents sitting idle at one site while other sites are overloaded?

Avaya Response:

Yes, the VoiceCon requirement is for 115 active and 200 configurable agents in support of 15 agent groups/skills but is not specific as to where these agents will be located, Since the Avaya Solution is a single system call loads will be balanced across the single instance of Avaya Communication Manager with Call Center Software - Elite active.

Since these agents are all off the same Communication Manager instance there is no need to implement Avaya Virtual Routing [Multi-Site Best Service Routing (BSR)] to load balance call routing and resource selection as all the 200 agents assigned to Skill #1 (Service) can be all equally considered for both call and agent selection.

For additional details please see Avaya Appendix 5 Avaya Contact Center Offers and Avaya Appendix 10 Avaya Call Center Software Call Handling Options Review for more details.

• Are all call center operations transparent across sites for ACD call routing, supervisory and reporting functions, telephony features, and any applications such as call recording?

Avaya Response:

Yes, Avaya’s proposed ACD solutions features transparent call center operations not only across sites for ACD call routing, supervisory and reporting functions, telephony features, and any applications such as call recording but for across all agents as they become staffed, regardless of their geographic location.

• Can your system guarantee Quality of Service (QoS) at the PSTN level for the voice channel for remote agents, and if so how is this done?

Avaya Response:

Yes, the IP Media Processor provides the transmission of voice data over an IP network. This enables support of applications that comply with H.323-v2 protocols. It also reduces perport costs and improves quality through its dynamic jitter buffers. Additionally, it performs echo cancellation, silence suppression, Dual Tone Multi-Frequency (DTMF) detection, and conferencing. This circuit pack provides Voice-over-Internet Protocol (VoIP) for Avaya IP Agent. It is used for the Road Warrior, IP Telephone, and Avaya Telephone-IP configurations where a VoIP connection is made to the Avaya Server that is driving the Avaya Communication Manager.




Voice over IP (VoIP) applications put severe constraints on the amount of end-to-end transfer delay of the voice signal and routing. If these constraints are not met, users complain of garbled or degraded voice quality, gaps, and pops. Due to the vagaries of human voice perception, VoIP applications can afford to randomly lose a few voice packets and the user can still understand the conversation. However, if voice packets are delayed or systematically lost, the destination experiences a momentary loss of sound, often with some unpleasing artifacts like clicks or pops.

For a more detailed discussion of packet delay and loss, we would direct the interested reader to the section on “Voice Quality Network Requirements” within the document available from [email protected] entitled Avaya Applications Solutions IP Telephony Deployment Guide. So as to mitigate potential surprises, Avaya recommends a network assessment that measures and solves latency issues before implementing VoIP solutions. Some of the factors causing voice degradation are packet delay and loss, echo, and transcoding.

• Are remote agents measured, service observed, recorded the same as local agents? Can they be members of the same ACD group, queue, split/skill as local agents?

Avaya Response:

Yes, Avaya’s proposed ACD solutions allow your staffed remote agents to be measured and service observed to be recorded in exactly the same manner as your staffed local agents.

9.0.4 Redundancy VoiceCon requires a high level of service availability for its contact center solution.


What levels of redundancy are embedded into your ACD solution design? Be specific as to ACD call control and routing functions, switched connections, announcements, and MIS reporting capabilities. Identify if redundancy is based on fully duplicated or load sharing hardware or software elements.

Avaya Response:

Comply and Avaya’s proposed ACD solution, comprised of an Avaya Communication Manager (CM) 5.0 with Avaya Call Center Software – Elite 5.0, will be configured to support a high level of service availability. Service availability is said by many to equate to reliability which is typically expressed in term of some quantity of 9’s. This means that if something is reliable 99% of the time then its reliability can be said to be 2 – 9’s. Three 9’s would equate to 99.9% and Avaya defines “high” as 99.99% up time.

So if we assume that VoiceCon needs their solution to be available for their use all 24 hours per day, 7 days per week or 8,766 hours (365.25 days) per year then we can say that 99.99% availability translates to 8.765 hours of non-availability per year of some sort.

Since a high level of reliability has been requested as opposed to a critical level of reliability Avaya will be relying upon Local Survivable Processing (LSP) to call control and routing functions are well cared for. While announcement reliability strategies vary, what many Avaya call center customers chose to do is to have multiple copies of the same announcement located either in different vVals located on the multiple Avaya G450 Media Gateways in their configuration.




Within the proposed solution for callers to the ACD agents, we would make use of Gateway-Server - “registered/unregistered” conditional commands within vectoring. These conditionals have be added to the “goto step/vector” command to allow alternate routing of calls if the H.248 Media Gateway or Port Network Gateway is not registered with the Avaya Server processing the call or if a backup server is processing the call (in survivability mode due to IP connectivity failure). This capability was added in Call Center Software 3.0 and not only intrinsically provides more robust call center availability via resource awareness but allows for better business continuance/disaster recovery planning.

These conditionals are used to determine if the Avaya Media Gateway is registered with a backup Avaya Server in survivability mode due to IP connectivity failure. We support three (3) conditionals (“media gateway”, “port network”, and “server”) that can be used with the goto step and/or goto vector commands to determine which server (set of “brains”) is processing the vector and which gateways are actually registered with that controlling server. The syntax of these relatively conditionals is as follows:

goto step if media-gateway [1 – x, “all” or “any”] [= or <>] registered

goto step if port-network [1 – x, “all” or “any”] [= or <>] registered

goto vector @step if media-gateway [1 – x, “all” or “any”] [= or <>] registered

goto vector @step if port-network [1 – x, “all” or “any”] [= or <>] registered

where:

media-gateway = H.248 media gateway (such as the G450),

port-network = a port network gateway (such as a G650)

x = the maximum number of gateways supported by the installed server platform

all = all of the equipped gateways

any = any one of the equipped gateways

Registered refers to the connectivity with the Communication Manager server currently processing the vector step for the call.

match on = registered means the specified gateway is registered with the server processing the vector step

match on <> registered means the specified gateway is not registered with the server processing the vector step

goto step ____ if server [=, <>] [‘main’, ‘ess’, ‘lsp’]

goto vector ____ @step __ if server [=, <>] [‘main’, ‘ess’, ‘lsp’]

Server refers to the server currently processing the vector step for the call. The server type is identified by the keyword ‘main’, ‘ess’ or ‘lsp’. The keyword ‘main’ refers to the main or primary server for the network/switch configuration. The keyword ‘ess’ refers to an Enterprise System Server (such as the S8500) that has been activated for the subnet to act as a backup server. The ‘lsp’ keyword refers to a Local Survivable Server, such as an S8300 or S8500 that has been activated for the subnet to act as a backup server.

Avaya’s Professional Services Organization (Consulting and System Integration) will be happy to assist in working with the VoiceCon Team to fine tune this process.




9.1 ACD Contact Call Center Parameters VoiceCon requires an ACD contact call center capability to support the following basic parameters:

• 115 active and 200 configurable agents with workstations

• 5 active and 10 configurable supervisors with workstations

• 15 agent groups/splits

• 50 integrated announcements with scripts

Confirm the proposed communications solution can satisfy these parameter requirements, and complete the following table.

Avaya Response:

Comply and Avaya’s proposed ACD solution can easily support 200 administered (configurable) agents with workstations, with 115 of them concurrent active simultaneously.

Proposed ACD System Parameters Maximum # Configurable Agents with workstations 1,000

Maximum # Active Agents with workstations 1,000

Maximum # Configurable Supervisors w/workstations 1,000

Maximum # Active Supervisors w/workstations 1,000

Maximum # Splits/Groups 2,000

Maximum # Trunk Groups 2,000

Maximum # ANI/DNIS numbers 20,000

Maximum # Call Delay Announcements 3,000

Maximum # Queue Slots 15,000

Standard MIS Reports 3,000

# Real Time 49

# Historical 56

# Exception 5




9.2 Basic ACD Features Vendor Response Requirement:

Confirm that the proposed ACD solution supports each of the listed features/functions, identifying any feature/function not supported:

The proposed ACD solution should include, at minimum, the following basic features:

• Agent mobility (e.g. login at any terminal);

Avaya Response:

Comply. The proposed ACD solution from Avaya supports the ability for administered agent using either a “hard” or “soft” phone to login from anywhere in the world to receive inbound ACD calls and place outbound calls.

• Multiple Agent Groups

Avaya Response:

Comply. The proposed ACD solution from Avaya support not only supports multiple call queuing but up to 2,000 skills (hunt groups / agent groups) in a single Avaya Communication Manager instance with the ability for a single skill to be as large as 1,000 members with the proposed Avaya S8500C Server. Agents can be taking calls in as many as 60 skills each.

• Call Flow

Avaya Response:

Comply. The proposed ACD solution from Avaya supports call control, screening and routing based upon Incoming Trunk Group, Dialed Number Identification Service (DNIS), Automatic Number Identification (ANI), Information Indicator (II) Digits (caller line origination type if delivered along with ANI), Calling Line Identification (CLID), Call Volume, System Performance Criteria, Queue Level Priorities, Skill Level Priorities, Greatest Need, Caller Enter Digits from Call Prompting and more.

• Applications/Skills-based Routing

Avaya Response:

Comply. The proposed ACD solution from Avaya supports applications/skills-based routing. Avaya’s term for “skills based routing” is Expert Agent Selection (EAS) which is part of the Avaya Call Center Software: Elite standard with Avaya Communication Manager 5.0.

• Priority Queuing

Avaya Response:

Comply and with Call Vectoring active in the proposed ACD solution from Avaya there are actually six levels of queue priorities downward from “Redirection On No Answer (RONA)” to “Direct Agent” to “Top” to “High” to “Medium” to “Low” which are applied regardless of when the inbound ACD call to the skill’s queue actually arrives. When Skill Level is administered for use by agent under call surplus conditions then as staffed agents for calls for a given skill become available (idle) their calls get delivered based upon skill level downward from 1 the most primary to 16 the least primary.




• Call Overflow and Interflow

Avaya Response:

Comply.

• Redirect on No Answer

Avaya Response:

Comply and the most popular method of applying Redirection on No Answer (RONA) is to RONA to a Vector Directory Number (VDN) which facilitates separate MIS tracking from a historical, real-time and exceptions tracking perspective.

• Predictive Overflow

Avaya Response:

Comply.

• Recorded Announcements

Avaya Response:

Comply.

• Music between Recordings

Avaya Response:

Comply.

• Dial out of Queue

Avaya Response:

Comply.

• Work codes

Avaya Response:

Comply. Event (Stroke) Counts are supported as well as Call Work Codes.

• Work timers

Avaya Response:

Comply.

• Make Agent Position Busy

Avaya Response:

Comply.

• Agent Help Request to Supervisor

Avaya Response:

Comply.

• Silent Monitoring – Split Monitoring: Supervisor & Agent

Avaya Response:

Comply but since EAS is active we would call this “Silent Monitoring – Skill Monitoring” for both Supervisor and Agent since with EAS the term “skill” is used instead of “split.”




• Silent Monitoring – Split Monitoring: Supervisor, Agent and Caller

Avaya Response:

Comply but since EAS is active we would call this “Silent Monitoring – Skill Monitoring” for both Supervisor and Agent since with EAS the term “skill” is used instead of “split.”

• Threshold Alerting

Avaya Response:

Comply.

• Queue status

Avaya Response:

Comply.

9.3 Call Flow At minimum the proposed ACD contact call center solution must be able to provide call control, screening, and routing based on:

• Incoming Trunk Group

• ANI, DNIS, or CLID

• Call Volume

• System Performance Criteria

• Priority Queuing

• Call Prompts


Confirm that the proposed ACD solution supports each of the listed call control criteria and briefly describe how supervisors/administrators create and develop scripts for incoming call flow operations.

Avaya Response:

Comply with all Call Flow requirement listed above and call vectoring is used as the straight forward call flow scripting as described in the final section of Avaya Appendix 5 - Contact Center Offers.

9.3.1 Routing & Queuing Vendor Response Requirement:

Confirm the proposed ACD solutions can support, at a minimum, each of the following call flow routing & queuing decision criteria, identifying any not supported.

• First In/First Out (FIFO)

• Time of day (TOD) / Day of week (DOW) / Day of year (DOY)

• ANI/DNIS/CLID

• Originating call by voice terminal type, i.e., cell phone or payphone

• Call prompt response

• Number of calls in queue

• Abandoned Calls




• Longest held call in queue

• Estimated wait time

• Available agents (number, skill)

• Agent idle time

• Agent handle times

• Caller directed routing

Avaya Response:

Comply with all Routing and Queuing requirements listed above with one note on Originating call by voice terminal type when Information Indicator (II) Digits are passed along with the Automatic Number Identification (ANI) with the arriving inbound call Avaya can route based on administration of the Communication Manager software. This means that II Digits are not capable of being used if they do not arrive along with the ANI (calling party’s telephone number).

9.3.2 Agent Skills Vendor Response Requirement:

Confirm that the proposed ACD solution supports skill-based routing, and identify:

• Total number of programmable skills;

Avaya Response:

Comply. The Avaya S8500C Server supports 2,000 skills per system and up to 60 skills per agent. Avaya’s proposed ACD solution supports Expert Agent Selection (EAS) as part of Avaya Call Center Software Elite. EAS is Avaya’s term for what the industry calls “skills based routing.”

• Number of programmable and active skills per agent;

Avaya Response:

Comply. The Avaya S8500C Server supports 2,000 skills per system and up to 60 skills per agent.

• Number of programmable and active multiple group assignments;

Avaya Response:

Comply. Although the proposed Avaya ACD solution can feature both agent group assignments and skill (hunt group) assignments Avaya, recognizes that inbound calls are funneled to skills (hunt groups) rather than to agent groups which are used for reporting purposes. The total number of programmable skills ranges from 99 if opting to deploy with an Avaya S8300 Server to 2,000 if you opt to deploy with either the of the Avaya S8500C, S8710, S8720, S8720XL or S8730 Servers.




• If an agent can add/delete their skill assignments;

Avaya Response:

Yes with clarification, an agent can add/delete all of their traditional skill (hunt group) assignments. However in the event that they are using Direct Agent, that one first administered skill cannot be removed since this is typically primary skill.

9.3.3 Customer Preference Vendor Response Requirement:

Confirm the proposed ACD solution supports customer preference call routing and queuing capabilities, and briefly describe the process how incoming callers can control their call flow.

Avaya Response:

Comply. The proposed ACD solution from Avaya supports customer preference call routing and queuing. The process by which an incoming caller can control their call flow generates a far less expansive answer than if we opted to respond with a description of the myriad ways in which callers inbound call flows can be created to reflect customer preference call routing and queuing capabilities. Callers control how their call will get handled within the proposed ACD solution based upon the typically 10-digit telephone number they called (toll-free or not), as well as the type of telephone line they are using to place the call (cellular or not), in addition to the day-of-the year (holiday or not), day-of- week, time-of-day and their response to opportunities to react to call prompting, inclusive of the use of caller entered digits as well as the number they are calling from in addition any data-directed call handling that might be additionally applied to inbound calls to the VoiceCon selected type of telephone.

9.4 Call Processing Functions Provide brief answers to each of the following questions regarding the proposed ACD solution call processing features and operations.

9.4.1 Describe how call control and agent handling methods can be administered locally and changed on demand if necessary in response to system activity.

Avaya Response:

The proposed ACD solution from Avaya features Expert Agent Selection (EAS) and supports the ability for an inbound ACD call that can arrive against any of up to 20,000 Vector Directory Numbers (VDNs) to potentially as many 2,000 skills that either have multiple skill queuing rules applied (calls can be queued to 3 skills simultaneously). Alternatively to queue for a given skill if agents are unavailable agents in that skill at one of six (6) different queue level priorities, in this order from Redirection On No Answer (RONA) – Direct Agent -Top – High – Medium – Low for the skill (hunt group).

This would be “call control” and the up to 2,000 vectors of up to 99 steps each that are used to control routing are accessed via supervisors using Avaya CMS Supervisor at their desks. The actual election of which inbound call to deliver to an available agent is capable of being influenced not only based upon priority of the calls in queue but also based upon agent skill levels, call handling preferences, and routing algorithms active. These include the use of Service Level Maximizer (SLM) which is included along with Skill Level and Greatest Need as Call Preference settings (administered by agent) applicable under Call Selection conditions (calls in queue) as well as an alternative to the use of the typically deployed agent selection options (administered per skill) of Uniform Call Distribution – Most Idle Agent (UCD-MIA), Uniform Call Distribution – Least Occupied Agent (UCD-LOA) , Expert Agent Distribution –




Most Idle Agent (EAD-MIA), and EAD-LOA or the additive to our optional Business Advocate. Avaya has the ability to tailor a sophisticated call prioritization scheme to help you meet your specific business goals.

The call control methods are what occurs during call surplus conditions where the administered settings for call handling preferences such as Greatest Need, Service Level and Service Level Maximizer settings made via the SAT come to bear to deliver queued calls to staffed agents. Agent handling methods are what occurs during agent surplus conditions when agent selection options, also administered by SAT, such as UCD-MIA, UCD-LOA, EAD-MIA, EAD-LOA and Service Level Maximizer are used to move agents to idle queues.

For additional details please see Avaya Appendix 8 Avaya Call Center Vectoring and Avaya Appendix 10 Avaya Call Center Software Call Handling Options Review for more details.

9.4.2 How many priority levels can be assigned incoming calls? Can priority level be changed while it is in queue based on system factors, i.e., time in queue, available agents, etc.?

Avaya Response:

Yes, the proposed ACD solution from Avaya supports the ability for the priority levels of queued calls serviced downward from Top to High to Medium to Low for a given skill to be easily changed based upon a myriad of different conditional decisions within vectoring that can include number of staffed, number of available agent, time-in-queue, etc.

9.4.3 How many callers can concurrently listen to a particular ACD recorded announcement? Is the number based on origin of the announcement, i.e., internal or external to ACD system?

Avaya Response:

The proposed ACD solution from Avaya supports the ability for 1,000 separate callers to concurrently listen to one specific ACD recorded announcement and/or audio source.

The supported quantity (number) of callers who can concurrently listen to one specific ACD recorded announcement and/or audio source is not based upon the origin of the announcement.

9.4.4 Can announcements played for a caller be defined as “uninterruptible” even when agents are available to handle the incoming call?

Avaya Response:

Yes this is easily accomplished via the use of call vectoring and in addition, if a call center is already using VDN of Origin Announcement (VOA) these VOA type announcements (typically administered as no longer then 1.2 seconds in duration) will always play to the agent (they cannot be heard by the caller) prior to the inbound ACD call actually being connected to the staffed agent.

9.4.5 Describe all available and standard automated prompt features that a caller would have to listen and react to for call screening and routing procedures.

Avaya Response:

The Avaya proposed solution for VoiceCon 08 supports a maximum of 3,000 announcements and comes with 2,000 vectors of 99 steps each and support Call Vectoring – Prompting which enables the following considerations:




digits >, >=, <>, =, <, <= Threshold value or string of digits: 1-16, wildcards (?, +), A-Z, AA-ZZ, V1-V9

<>, = none = meet-me-access in table 1-100, A-Z, AA-ZZ, V1-V9 not-in table

Automated attendant example

This example scenario shows the use of Automated Attendant, which is one of the applications that can be supported by the Call Prompting feature. Automated Attendant allows the caller to “Voice Con 08” to enter either the extension of the party or the EAS Agent Login ID of the agent that the caller wants to reach. Depending on the parameters established, the user can enter up to 16 digits from a touchtone telephone.

Automated Attendant is usually used by call centers that do not have DID trunks and whose callers know the extension of the people they are calling. Because it reduces the need for live attendants, its use can reduce ongoing call center costs while placing the “call routing, menu prompting, control in the hands of the calling party.”

The following example shows an example of a vector that implements Automated Attendant.

Example application - automated attendant 1. wait-time 0 seconds hearing ring back 2. collect 5 digits after announcement 30001 [You have reached VoiceCon in Orlando.

Please dial a 5-digit extension or wait for the attendant.] 3. route-to digits with coverage y

Step 1 of this vector contains the wait-time command, which is placed before the collect digits command in step 2 to provide the caller with ring back in the event that a TTR (Touch Tone Receiver) is not immediately available. A TTR must be connected in order for the collect digits command to take effect. Once a TTR is connected, the caller is prompted to enter the destination extension of the party he or she wants to reach (step 2). The collect digits command in step 2 collects the digits. Thereafter, the route-to digits command in step 3 attempts to route the inbound call to the desired destination. If the route-to digits command fails because the caller fails to enter any digits, or because the digits entered do not comprise a valid extension within your dial plan, then the route-to number command in step 4 routes the call to the attendant (operator) if you have one administered. However, as long as the destination is a valid extension, the route-to digits command succeeds, coverage applies, and vector processing terminates. Note that even if the desired destination is busy, vector processing terminates because coverage call processing takes effect.

Data in/voice answer and data/message collection example

This example involves a mutual fund company that is open 24 hours a day, 7 days a week. All incoming calls are directed to a single VDN (soft number extension) that maps to a main vector. The main vector presents a menu of options to the calling party, and the vector also uses Call Prompting to determine the desired service. Three services are offered:

1. New accounts enables the customer to open a new account.

2. Account inquiries enables the customer to make inquiries concerning their account.

3. Net asset values enables the customer to hear information concerning the net asset values of the company’s funds.




If the caller selects account inquiries, they are then prompted to input their account number before being answered by an agent. The agent can use the CALLR-INFO button to display this number and if the agent happens to be using two-line display telephone then the entered account number is automatically displayed on the second line.

This example uses three other applications that can be supported by the Call Prompting feature:

¾ Data In/Voice Answer (DIVA) allows a caller to receive information on a topic that they opt to select at the prompt. The caller selects the desired topic by entering the appropriate digits

¾ Data Collection provides a method of collecting digits from a caller. The requested digits comprise an official number of some sorts. For example, a numeric account number or perhaps a telephone number that is linked to the caller’s account so as to they help the system process the call more efficiently.

¾ Message Collection allows the caller to leave a recorded message instead of waiting for the call to be answered.

The following vectors (10, 11, 12 and 13) illustrate how the mutual fund company handles telephone calls.

Example application - mutual fund company

VDN (extension=1030 name="ABC Inv" vector=10 display override="y")

Vector 10 1. wait-time 0 secs hearing ring back 2. collect 1 digits after announcement 3531

[Thank you for calling ABC Investments. If you wish to open a new account, please dial 1. If you wish to make an account inquiry, please dial 2.If you wish to know the current net asset values of our funds, please dial 3.]

3. route-to number 1031 with cov y if digit = 1 4. route-to number 1032 with cov y if digit = 2 5. route-to number 1033 with cov y if digit = 3 6. route-to number 0 with cov n if unconditionally 7. disconnect after announcement none

VDN (extension=1031 name="New Account" vector=11)

Vector 11

1. goto step 5 if calls-queued in skill 1 > 19 2. queue-to skill 1 pri t 3. announcement 3535 4. wait-time 10 secs hearing music 5. collect 1 digit after announcement 4020 [We’re sorry. All of our representatives are busy

at the moment. If you’d like to leave your name and telephone number so that we can get back to you, dial 1.]

6. goto step 10 if digit = 1 7. announcement 3537 8. wait time 50 secs hearing music 9. goto step 6 if unconditionally 10.messaging split 5 for extension 4000 11.Announcement 3538 [We’re sorry; we cannot take your message at this time. You may

continue to hold, or you can call back later.]




12. goto step 4 if unconditionally

VDN (extension=1032 name="Account Inq" vector=12)

Vector 12: 1. wait-time 0 secs hearing ring back 2. collect 6 digits after announcement 3533 3. [Please enter your 6-digit account number.] 4. goto step 7 if calls-queued in skill 1 > 19 5. queue-to skill 1 pri m 6. announcement 3535 7. wait-time 60 secs hearing music 8. collect 1 digits after announcement 4020 9. [We’re sorry. All of our representatives are busy at the moment. If you’d like to leave

your name and telephone number so that we can get back to you, dial 1.] 10.goto step 12 if digit = 1 11. announcement 3537 12.wait time 50 secs hearing music 13.goto step 8 if unconditionally 14.messaging skill 5 for extension 4000 15. announcement 3538 [We’re sorry, we cannot take your message at this time. You may

continue to hold, or you can call back later.] 16.goto step 4 if unconditionally

VDN (extension=1033 Name="Net Asset Val" Vector=13)

Vector 13:

1. disconnect after announcement 3534

[The net asset values of our funds at the close of the market on Wednesday, May 15 were as follows: ABC Growth.....33.21.....up 33 cents; ABC High Yield.....11.48.....down 3 cents.]

When the call is placed, vector processing begins in vector 10, which is considered the “main” vector since that is where the VDN (vector directory number) initially enters the Avaya Call Center Software - Elite. Step 1 of the vector contains the wait-time command, which is placed before the collect digits command in step 2 to provide the caller with feedback in the event that a tone detector is not immediately available. Once a tone detector is connected, the collect digits command provides an announcement that requests the caller to enter 1, 2, or 3, depending upon the service desired. If the caller enters a digit other than 1, 2, or 3 mentioned, or if the caller fails to enter any digits within 10 seconds (this is an administrable timeout), then the command fails and the call is routed to the attendant (step 6 of Vector 11). If the caller enters 1, 2, or 3 within 10 seconds, then the call is routed to the vector specified in the appropriate route-to number command, which appears in steps 3, 4, and 5 of Vector 11.

For instance, assume that, when prompted, the caller enters 3 because they want to learn about the net asset values of the company’s funds. In such a case, the route-to number commands in step 3 and in step 4 fail, because in each case, the digit that is tested for in the condition portion of the command is not 3. However, the route-to number command in step 5 succeeds because the digit that is tested for matches the one entered by the caller. Accordingly, the call is routed to VDN extension 1033, and vector processing continues in vector 13.




The announcement command in step 1 of vector 13 provides the caller with the information on net asset values and then disconnects the call.

The process just described, whereby the caller receives information as a result of making a request at the prompt, is an example of the Data In/Voice Answer (DIVA) application.

Returning to the main vector (Vector 10) let’s suppose that another caller wants to make an inquiry into their account, and the caller enters 2 when prompted. In such a case, and step 3 fails, but step 4 succeeds. Accordingly, the call is routed to VDN extension 1032, and vector processing continues in vector 12.

The collect digits command in step 2 of vector 12 first requests the caller to enter their 6-digit account number. The command then collects the digits that are entered by the caller. Whether or not the caller correctly enters the digits, the queue-to split command in step 4 queues the call. If an agent does not immediately answer the call, the standard announcement is provided in step 5 and, if necessary, a delay is provided in step 6. The announcement in step 7 provides the caller with the option of leaving a message instead of having their call wait in queue. The caller is instructed to enter 1 if they want to leave a recorded message. If the caller does not enter 1, the goto step command in step 8 fails, and an announcement-wait cycle is implemented by steps 9, 10, and 11 until the call is answered or abandoned. If the caller does enter 1 within 10 seconds, step 8 passes control to step 12. The messaging split command in step 12 attempts to connect the caller to an AUDIX or message center split so that the caller can leave a message. If the connection is made, the caller first hears ring back and can then leave a message. If the connection is not made, the step is unsuccessful, and step 13 provides an announcement that indicates that a connection could not be made. Thereafter, the goto step command in step 14 sends call control back to step 6, which leads the caller back into the steps to leave a message.

The process that was just described, whereby the caller, when prompted, enters digits that comprise an official number (an account number, in this case), is an example of the Data Collection application. If the agent has a CALLR-INFO button administered on the set or IP Agent, or if they happen to be using a phone with a two-line display, the agent can see the digits that are entered by the caller. As a result, the agent need not request the calling party’s account number.

Finally, suppose that a third caller wants to open an account and that he or she enters 1 when prompted in the main vector. In this case, step 3 of the main vector is successful. Accordingly, the call is routed to VDN extension 1031, and vector processing continues in vector 11.

In step 2 of vector 11, the call is queued to the main split. Thereafter, if necessary, step 3 provides the appropriate announcement, and step 4 provides a delay period. The announcement in step 5 provides the caller with the option of leaving a recorded message instead of having to wait in queue. This is an example of the Message Collection application. The caller is instructed to enter 1 if they want to leave a recorded message. If the caller does not enter 1, the goto step command in step 6 fails, and an announcement-wait cycle is implemented by steps 7, 8, and 9 until the call is answered or abandoned. If the caller does enter 1 within 10 seconds, step 6 passes control to step 10.




The messaging split command in step 10 attempts to connect the caller to an AUDIX or message center split so that the caller can leave a message. If the connection is made, the caller first hears ring back and can then leave a message. If the connection is not made, the step is unsuccessful, and step 11 provides an announcement that indicates that a connection could not be made. Thereafter, the goto step command in step 12 sends call control back to step 4, which leads the caller back into the steps to leave a message.

9.4.6 Can callers maintain their position in queue while interacting with an IVR?

Avaya Response:

Yes the proposed ACD solution from Avaya supports callers maintaining their queue position.

9.4.7 Describe any other unique call routing features available on your ACD system that you believe VoiceCon would be interested in knowing about.

Avaya Response:

There are two unique call routing products that VoiceCon can use in their potential Avaya contact center solution that would be of tremendous value-add. The first is included as standard with Avaya Call Center Software-Elite with active Expert Agent Selection (EAS). In the proposed VoiceCon solution, having the use of Single-Site Best Service Routing (BSR) can make a difference to your customer and/or product/solution segmentation call handling strategies. Because BSR is more then just routing, VoiceCon can make use of BSR (Best Service Routing) Available Agent Strategy by Vector Directory Number (VDN) which will provides unique granularity of call delivery under agent surplus conditions. For each of the up to 20,000 VDNs that VoiceCon can have taking inbound ACD calls for you can administer one of 5 unique BSR Available Agent Strategies. Ideally the BSR Available Agent Strategy assigned to a VDN should match the agent selection method used in the skill considered by a BSR application we’d probably recommend the use of Expert Agent Distribution – Least Occupied Agent so that the next selected staffed agent will be that available (idle) agent with the lowest percentage of work time (meaning least agent occupancy) since login as compared to other available agents having the exact skill level assignment. However if the inbound calls connected to some VDNs are more important than what the EAS-LOA algorithm provides than we’d recommend using the BSR Available Agent Strategy of First Found against those VDNs so as to assure that the inbound ACD call will be given to the first available (idle) agent regardless of their call handling expertise.

The second unique call routing product is optional and we will just recap one capability of the optional Business Advocate for Avaya Communication Manager software which requires that Expert Agent Selection (EAS) be active and that the EAS option of Service Level Maximizer (SLM) be inactive. This capability is called Dynamic Queue Position (DQP) or Service Objective by Vector Directory Number (VDN) and if active would allow VoiceCon to administer a separate targeted service level for each of the up to 20,000 VDNs that they proposed solution supports. This makes it far easier to provide callers with differentiated service under call surplus conditions. DQP takes precedence over Service Objective by Skill (Hunt Group) and offers optimal “answer X% in Y seconds” routing granularity. If you assume that one (1) ring cycle equates to six (6) seconds then just work with your line of business owners to obtain a listing from them as to “how many rings should callers to these respective VDNs hear before being answered.“




9.5 Supervisor Functions The proposed ACD solution must be able to support supervisor positions to monitor and assist call agent positions, monitor and review system performance, and administer ACD functions and operations.

Provide answers to each of the following supervisor function requirements.

9.5.1 Describe how a supervisor can remotely monitor an agent. Indicate if the agent is notified by the system if they are being monitored by a supervisor, and specify the type of notification signal.

Avaya Response:

Comply. The proposed ACD solution from Avaya supports the ability for supervisory positions to monitor and assist call agent positions, monitor and review system performance, and administer ACD functions and operations remotely and the agent is totally unaware their they are being monitored. Manual intervention via dial intercom could be used by the supervisor to the staffed agent to alert them that they are being monitored.

9.5.2 Can a supervisor assist an agent during an active call, and if so can the agent received prior notification this is about to occur?

Avaya Response:

The proposed ACD solution from Avaya supports the ability for a staffed supervisor to assist a staffed agent during an active call and the agent who is requesting supervisory assistance will be notified as they are requested. The “standard” most prevalent manner for an agent to request supervisory assistance from a skill supervisor would have a staffed agent who may or may not happen to have an active ACD call in progress would presses an Assist button, either administered on their physical set or perhaps via either IP Agent Shared Control or IP Agent if remotely located or places the call in progress on hold and dials the “Assist Facility Access Code (FAC)” plus the skill number. The agent must be logged into a skill that they are requesting assistance in which will not be as issue with active Expert Agent Selection as part of Avaya Call Center Software Elite. Assist generates 3-burst ringing at the supervisor’s station. If a skill supervisor is

not assigned, the agent receives intercept tone. If an Attendant Console is involved an “attendant console user” should press the Start button before pressing the Assist button. This allows them to later transfer the call.

When a “supervisor assist call” rings into a supervisor it rings like a priority call at the supervisor’s set. When the staffed agent presses the Assist button, the following happens:

1. If the agent happens to be active on an ACD call, the ACD call is automatically placed on hold and a “supervisor agent assist” call is placed to the skill. If the agent is not active on an ACD call, then a call is automatically placed to the supervisor.

2. Both an Avaya Call Management System (CMS) and/or BCMS would be notified of the request and the called supervisor’s display shows that the call is a request for assistance. This rings like a priority call at the supervisor’s telephone set.

3. The caller will hear either silence or music on hold.




4. After the staffed agent who has requested assistance has talked to the staffed supervisor, the agent can drop the assist call and return to the ACD call, set up a conference call with the supervisor and the calling party, or instead transfer the call to the supervisor. When the agent places the call with the customer they are taking on hold and next dials the Assist Facility Access Code (FAC) plus the skill number, the Avaya Communication Manager handles the request as if the agent pressed the Assist button, except that the “Assist call” does not follow the supervisor’s administered call coverage path.

Because the use of the Assist FAC does not trigger the ability for a “supervisor assist call” to be subsequently dealt with by following the supervisor’s administered call coverage path in the event that the requested supervisor is unavailable this method is almost never deployed

in a customer environment. Instead configurations go with an “Assist” button approach.

Avaya estimates that out of 100% of those designs that rely upon use of the Assist Button to reach a supervisor that roughly 80% of them will go ahead a label an Assist Button on the agent’s phone but actually have it programmed so that when this button is depressed by a staffed agent who needs supervisory assistance that call goes not to a specific supervisor to rather to a particular Vector Directory Number (VDN) that can in turn be sent anywhere. The beauty of this alternative approach frequently, recommended by Avaya Consulting and System Integration, is that calls to VDNs are measured by the Avaya Call Management System (CMS) and can therefore be easily tracked in terms of their frequency, duration and disposition. Given the goals of a

particular business unit and the expanding and contracting ratios of “supervisory support staff” to “staffed agents” at different times during a taking day but having “supervisory assist calls” arriving at one or more separate groups of “people who can provide general supervisory assistance” may be far more cost effective and efficient that in a center where the ratio of agents to supervisors is 12 to 1, simply taking the chance that the supervisor who can assist is available (idle) and ready to assist at any time.

9.5.3 Identify and briefly describe reports that are available to a supervisor via their PC monitor. Attach samples of available display screenshots.

Avaya Response:

Real-time reports available to a staffed VoiceCon 08 Supervisor via their PC monitor are delivered via Avaya CMS Supervisor and have been recapped in Avaya Appendix 6 – Avaya CMS Reports.

Avaya provides “agent focused real-time reports” such as:

¾ Agent Group Reports,

¾ Agent Report, and Agent Graphical Information

Avaya provides “queue/agent focused real-time reports” focused such as:

¾ Queue/Top Agent Status Report,

¾ Queue/Agent Status Report

¾ Queue/Agent Summary Report.

Avaya provides “trunk/trunk group focused real-time reports” such as Trunk Group Report.

Avaya provides “vector focused real-time reports” such as Vector Report.




Avaya provides “vector directory number (VDN) focused real-time reports” such as VDN Call Profile Report, VDN Graphical Call Profile Report, VDN/Skill Preference Report, and VDN Report.

Avaya provides “skill focused reports” such as Agent Status by Location Report, Call Profile Report, Skill Graphical Active Agents Report, Skill Graphical Allocated Agents Report, Skill Graphical AUX Agents Report, Skill Graphical AUX Top Agents Report, Skill Graphical Call Profile Report, Skill Graphical EWT (Expected Wait Time) Report, Skill Graphical Queue Report, Skill Graphical Skill Overload Report, Skill Graphical Staffing Profile Report, Skill Graphical Status Report, Skill Graphical Status Report, Skill Graphical Top Skill Status Report, Skill Reserve1 AUX Agents Report, Skill Reserve2 AUX Agents Report, Skill AUX Report, Skill Status Report, Skill Top Agent Report, Skill Report, Skill by Location Report, and Skill Top Agent Status Report.

Avaya provides “other focused reports” such as Event Count Summary Report, Multi-ACD Report, and Multi-ACD Top Agent Report.

In addition, real-time reports include “drill down focused reports” such as Reserve1 AUX Agent Report, Reserve2 AUX Agent Report, Reserve1 Work State Report, Reserve2 Work State, and Work State Report.

The included Historical and Integrated (these are combined Historical with Real-Time) Reports cover the same areas.

Please see Avaya Appendix 6 for a recap of the majority of these reports.

9.5.4 How often is data updated on supervisor monitor display?

Avaya Response:

The proposed ACD solution from Avaya supports a wide variety of data update settings to supervisor monitor displays with the shortest possible update being 3 seconds in duration.

9.5.5 Describe the various display screens a supervisor would have access to for real time management operations. Include an explanation for each screen field. Specify if any of the following agent performance metrics are displayed: service level; quantity and time of calls in queue; average speed to answer; number of agents staffed to handle calls; and identity any other available performance monitoring capabilities.

Avaya Response:

The Avaya proposed solution for VoiceCon does support displaying the following agent performance metrics (see our response to 9.5.3) and comments regarding agent focused real-time reports, queue/agent focused real-time reports and skill focused reports):

¾ Service Level

¾ Quantity and Time of Calls in Queue

¾ Average Speed of Answer

¾ Number of Agents Staffed




How your VoiceCon Supervisor will access reports means the way you select, run, edit, and view a report is the same. VoiceCon will find the appropriate reports under the following menus and tabs:

¾ Commands | Reports | Real-Time, Historical, or Integrated tab

¾ Commands | Dictionary | Reports tab

¾ Commands | Exceptions | Reports tab

¾ Commands | Agent Administration | Reports tab

¾ Commands | Call Center Administration | Reports tab

¾ Tools | Maintenance | Reports tab

If VoiceCon uses Report Designer and wants to easily generate a new customized report, select the Report Wizard button in the Controller toolbar. Use the Browser Dates window to select dates for a report. The Scripting feature lets you create a script to run a specified report or run a report and export the data on schedule. The report can run in the current Supervisor session and be displayed on the PC, or it can run in the background.

The Report menu has options that relate specifically to the report you are running.

The following table describes the action that each menu item performs:

Menu Item Action Restart Closes the current report output window and displays the input window for the

current report. This allows you to run the same report, changing inputs as needed. This is disabled for Drill-Down reports and if the report is an unsaved report created using the Report Wizard (this item is re-enabled when the report is saved).

Page Setup Displays the Page Setup window. For more information, refer to the Microsoft® Windows® documentation or Help.

Print Displays the Print window. When you select Print from the Report menu, the report that is active will be printed according to the options you choose in the Print window. If the report is a real-time report, only one refresh of the report will be printed. For more information, refer to the Windows documentation or Help.

Print Preview Displays the report so that you can see it before printing. Once you have selected Print Preview, you may print the report. For more information, refer to the Windows documentation or Help.




Menu Item Action Design If authorized to use the Report Designer, you can access the Report Designer

Design Mode window from any report using this menu item. When you select Design from the Report menu, the report that is currently running remains visible, but you are placed in the Design Mode, where you can edit the report. For more information on the Design Mode, see the Avaya Call Management System Supervisor Report Designer user guide. The Design menu item will be grayed out if you are currently running a drill-down report.

Save Saves the report that you edited. Save as Saves the report you edited under a different file name.

Save as HTML Displays the Save as HTML window, which lets you save a snapshot of the report output as an HTML file. Selecting the Script button in this window displays the Save as Script dialog box, which lets you create a script to run the specified report and save the output as an HTML file.

Script Displays the Save as Script dialog box, which enables you to create a script to run a specified report on schedule and display it on your computer. The script can be interactive or automatic.

Exit Closes the active report output window. The location of the report output window is saved when the report is exited, and that is where the report will display the next time you run it.

The information on a standard report window is organized as follows:




The graphical reports are displayed in charts:

There are two types of report layouts in Avaya CMS Supervisor:

¾ Tables to support the presentation of standard reports

¾ Charts to support the presentation of graphical reports

A report can consist of only table/tables, only chart/charts, or a combination of tables and charts. In addition to the tables and charts that make up the main body of reports, reports can include text and field names that are followed by single data items. Text and data fields are used on both table and chart reports.

Most standard reports (reports that are not prefaced by graphical are displayed in tables.)

Please review our Call Center Reports Avaya Appendix 6 for examples of reports.

9.5.6 Can a supervisor perform drag and click system management programming from their monitor, and if so briefly describe available management features and operations. Specify if a supervisor can perform real time reconfiguration of call flows and agent skills assignments.

Avaya Response:

Yes. A supervisor who is using Avaya Visual Vectors can perform drag and click system management programming from their monitor

The proposed ACD solution from Avaya supports the ability for a supervisor with the appropriate permissions to be able to perform real time reconfiguration of call flows and agent skills assignments.




9.5.7 Can supervisors log-out an agent remotely?

Avaya Response:

Yes. The proposed ACD solution from Avaya supports the ability for supervisors to remotely log-out a staffed (logged-in) agent.

9.5.8 Can supervisors force a group into night handling?

Avaya Response:

Yes. The proposed ACD solution from Avaya supports the ability for supervisors to force a group into night handling.

9.5.9 Can supervisors monitor and observe agents by agent ID? Can they listen and talk to an agent during a conversation: Can the entire customer experience be monitored, including announcements, music, etc.?

Avaya Response:

Yes. The proposed ACD solution from Avaya supports the ability to monitor/observe agents by agent ID on either a listen-only or a listen-talk basis so that they can listen and talk to an agent. The ability for the entire customer experience to be monitored inclusive of the announcements and music-on-delay that caller is experiencing is supported as well.

9.6 Agent Functions The proposed ACD solution must support a variety of agent functions. Provide answers to each of the following agent function requirements.

9.6.1 Describe the process for an agent to request supervisor assistance during an active call, and specify if the caller must be on hold or if the agent/supervisor can talk without the caller hearing the conversation. After the supervisor consultation can the caller be transferred or conferenced?

Avaya Response:

Please refer to the response to 9.5.2 for the request for supervisor assistance. The proposed ACD solution from Avaya supports the ability for the caller to be transferred or conferenced after the supervisor consultation.

9.6.2 What happens if a supervisor is not available when assistance is required?

Avaya Response:

The proposed ACD solution from Avaya supports call coverage for the staffed supervisor to ensure that call gets answered. Many Avaya call centers actually program the “Assist” button on their agent’s telephones to actually dial a unique Vector Directory Number (typically 3 to 7 digits in length) so as to reach a centralized help

9.6.3 Identify and provide a brief description of ACD system information (calls in queue, average time in queue, personal call handling statistics, et al) that can be retrieved and displayed on the agent desktop telephone instrument.

Avaya Response:

Staffed agents of the proposed ACD solution can receive real-time display information using VuStats on their physical telephone instrument the Avaya 9650 proposed or from Avaya IP Agent (Softphone) can accommodate this feature.




VuStats on the Agent Telephone Display

Avaya offers the VuStats feature for display of ACD statistics on the agent telephone instruments. VuStats is a convenient, cost effective way for call centers to measure results in real time. Anyone with a display-equipped voice terminal, including call center managers and non-ACD personnel, can use VuStats to view real-time or cumulative daily call center statistics. VuStats gives agents the power to judge their own performance and take steps to modify call handling skills to improve their productivity.

For example, agents who are staffed and happen to have VuStats can:

¾ View calls in queue and/or wait times to delay nonessential activities until call delays are acceptable

¾ View their time in Auxiliary (AUX) Work

¾ Compare their productivity with call center objectives or the performance of other agents and know when to step up the pace

¾ Keep track of their total cumulative performance for an entire day

¾ Be automatically notified by a flashing lamp when thresholds are reached for individuals and groups

Up to 50 different 40-character display formats (each with up to 10 fields of data) can be customized, thereby creating displays of information that are important to call center personnel. Thresholds can be defined on data items that will cause the VuStats lamp to flash when the displayed item exceeds a pre-defined threshold. All data is cumulative up to the current second, combining current interval and historical data. Most data can be cumulative for the entire day or for the most recent 24 hours or half-hours. Redisplay formats can be linked so the agents can step through a series of displays to view their progress against different measurements.

VuStats is included with Avaya Call Center Software: Elite and can be quite beneficial to a wide range of users with a call center.

VuStats Support on Avaya IP Agent Softphone

Agents and administrators can check the VuStats Monitor window for information on Contact Center operations, like the number of calls waiting for a particular split. The information is updated periodically, based on pre-defined refresh rates that users can select. Agents can automatically monitor one or more lines of VuStats information at the same time. The information will be presented in a window independent of the main window.

The user can administer the refresh rate for the VuStats screen. All the VuStats displays can be updated every 10, 20, 30, 60 and 120 seconds. The user can also administer the display interval per button between the display updates (i.e. the time the monitor waits to gather display information before moves to the next VuStats display). The display intervals can be 1, 3, 5 and 10 seconds. Intervals may need to be adjusted based upon network speed.




VuStats Monitor

PC Wallboards Keep Your Agents Informed (optional)

Wallboard is a Windows-based application that displays real-time statistical information on VDNs, skills or splits and agents in a marquee window. Installed on agent PCs, the scroll bar of information allows agents to closely track their personal work performance and the performance of their work group (skill or split). Statistical information is sent to the Wallboard application from the Interaction Data Server, which monitors VDNs, splits, skills and agent extensions, and then calculates statistics about all facets of a call.

This powerful feature is easy to administer. It eliminates the need for a separate expensive wallboard within a call center. Custom messages can be added with color and flash. For example, if an inbound call center wants to feature their TVs at a special customer price, that information could be flashed on the wallboard in red.

Wallboard can be used as an Agent extension or a standalone application. The following types of information can be displayed on the wallboard:

Wallboard VDN (Vector Directory Number) Information

¾ VDN names and numbers

¾ The quantity of calls waiting to be answered for the specified VDN

¾ The length of time the first call in the queue has been waiting (in seconds)

¾ The average length of time agents are talking to callers to this VDN (in seconds)

¾ The average length of time callers to this VDN are waiting before their call is answered (in secs)




¾ The number of calls to this VDN that have been abandoned

¾ The average length of time callers to this VDN are waiting before abandoning their calls (in seconds)

Wallboard Skill or Split Information

¾ Skill (or split if EAS is not active) names and extension numbers

¾ The quantity of calls waiting to be answered for the specified skill (or split if EAS is not active)

¾ The number of agents logged into the skill (or split if EAS is not active) that are available to take calls

¾ The number of agents logged into the skill (or split is not active) that are unavailable to take calls

¾ The average length of time agents logged into the skill (or split if EAS is not active) are talking to callers (in seconds)

¾ The average length of time callers to this skill (or split if EAS is not active) are waiting before their call is answered (in seconds)

¾ The number of calls made to the skill (or split if EAS is not active)

¾ The number of calls to the skill (or split if EAS is not active) that have been abandoned

¾ The average length of time callers to this skill (or split if EAS is not active) are waiting before abandoning their calls (in seconds)

Wallboard Agent Information

¾ Agent names, IDs and station numbers

¾ The current work mode of the specified agent

¾ The number of calls the agent takes per hour

¾ The average length of time the agent spends on a call (in seconds)

¾ The average length of time the agent spends in After Call Work (ACW) mode (in seconds)

¾ The average length of time the agent spends in Auxiliary (AUX) Work mode (in seconds)

¾ The agent's pending work mode

¾ The agent's pending reason code

¾ The last reason code the agent used

¾ The skill or split group the agent is logged into




9.6.4 Identify and provide a brief description of ACD system information and reports (calls in queue, average time in queue, personal call handling statistics, et al) that can be retrieved and displayed on the agent PC monitor. Attach samples of available display screenshots.

Avaya Response:

Contact Center statistics traditionally displayed using wallboards can now be presented on the agent’s workstation using the Avaya Desktop Wallboard application, which is an Avaya CSI Offering.

This is a popular alternative to the use of VuStats and BCMS on agent telephones. Many firms have realized that Contact Center performance is significantly increased by feeding back real time statistics to agents so that their current productivity can be compared with their objectives. The effectiveness of this motivational technique is enhanced further by displaying the information directly on the agent’s workstation in a format that highlights the current performance in comparison to their goals.

This is done by having a scrolling marquee installed on each agent workstation (PC) that real-time and historical statistical information from the Avaya Call Management System (CMS). It could also be fed by Avaya Operational Analyst (OA) but we have not included OA here. The format of the information that is displayed changes at configurable thresholds thereby highlighting current performance issues.

The Avaya Desktop Wallboard application features multiple marquees working in concert. The Contact Center Supervisor selects the statistical information to be displayed on the agent desktop such as number of calls in queue, average wait time, etc.). The Contact Center Supervisor can configure threshold goals for each data element using the Desktop Wallboard web-based administration application. This allows the data to be displayed in different text colors and for sounds to be played depending on the achievement in relation to the configured threshold goals.

The CMS marquee provides agents with a data feed from the Call Management System (CMS). The CMS marquee can be configured to use thresholds in monitoring CMS data and change the font color when an administered threshold is exceeded. The Message marquee provides a scrolling list of messages sent by administrators. Desktop Wallboard includes both client and server components. The server components must be installed on a dedicated client provided server. The client components are installed throughout the network.

The Avaya Desktop Wallboard solution component is installed on a customer provided dedicated Windows 2000 or 2003 server. This receives a predefined subset of data from the CMS or OA via the appropriate feed component over a TCP/IP Ethernet network. Client marquees are configured via a web-based administration tool to periodically receive specific items from the Server and display these to the agent. Administration is done through a web-based interface to specify messages and statistics that need to be distributed to agents. These messages and statistics are stored in a database on the Server. Some information is periodically pushed to the client and some is periodically requested from the server.




VuStats is included at no additional cost with the Avaya Call Center Software: Elite we are including in the proposed solution for Voice Con 08 and since up to 50 different 40-character display formats (each with up to 10 fields of data) can be customized, thereby creating displays of information that are important to call center personnel in intrinsically “business useful.” We want to work with your line of business owners to really understand how the agent’s will be using their desktop (real estate) in support of the 15 agent groups (skills) that have been specified prior to actually recommending the Avaya Desktop Wallboard application for use at VoiceCon.

Realistically the agents in the VoiceCon center will be best served by having those items that are within their control on the desktop telephone displays. This means Agent & Agent Extension Group Data Type information such as the logged in time they have spent not in either Direct Call Processing or in Post Call Processing as represented by [Shift aux time (19, all, default, non-default)], their own average talk as compared their agent group’s average talk time as represented by [Shift average ACD talk time], their own hold time as compared to their agent group’s average hold time as represented by [total hold time], and their own available (idle) time as compared to their agent group’s average available time as represented by [total available time].

VuStats supports displaying these ACD statistics on agent voice terminals: Trunk Group Data Types Item Number Trunk Group Data Type (in alphabetical order)

1 average incoming call time 2 average outgoing time 3 incoming abandoned calls 4 incoming calls 5 incoming usage 6 number (quantity) of trunks 7 outgoing calls 8 outgoing completed calls 9 outgoing usage 10 percent all trunks busy 11 percentage trunks maintenance busy 12 trunk group name 13 trunk group number 14 trunks in use 15 trunks in maintenance busy




Vector Directory Number (VDN) Data Types Item Number VDN Data Type (in alphabetical order)

1 acceptable service level 2 ACD calls 3 average ACD talk time 4 average speed of answer (ASA) 5 average time to abandon 6 calls abandoned 7 calls flowed out 8 calls forced busy or disconnected 9 calls offered 10 calls waiting 11 oldest call waiting 12 percent in service level 13 total ACD talk time 14 VDN extension 15 VDN name

Skill (and Split) Group Data Types Item Number Skill (and Split) Group Data Type (in alphabetical order)

1 acceptable service level 2 ACD calls 3 After call sessions 4 Agents available 5 Agents in after call work 6 Agents in AUX (1-9,all,default, non-default) 7 Agents in other 8 Agents on ACD calls 9 Agents on extension calls 10 Agents staffed 11 Average ACD talk time 12 Average after call work time 13 Average speed of answer (ASA) 14 Average time to abandon (ATA) 15 Call rate 16 Calls abandoned 17 Calls flowed in 18 Calls flowed out 19 Calls waiting 20 Oldest call waiting 21 Percent in service level 22 Split extension 23 Split name 24 Split number 25 Split objective 26 Total ACD talk time 27 Total after call work time 28 Total AUX time




Skill (and Split) Group Data Types Item Number Agent & Agent Extension Group Data Type (in alphabetical order)

1 Shift aux time (1-9, all, default, non-default) 2 Shift aux time reason code 3 Shift average ACD talk time 4 Split call rate 5 Split calls abandoned 6 Split calls flowed in 7 Total available time 8 Total hold time 9 Total staffed time

9.6.5 Describe how an agent is able to distinguish incoming calls as a new call, transferred call, or a call from a voice response system.

Avaya Response:

Since most call center agents are deployed with telephones featuring integrated displays it is the Vector Directory Number (VDN) name along with the zip tone from Auto-Answer that enables the staffed and answering VoiceCon 08 agent to realize that this is a new ACD call to be handled. There are visual alerts and the VDN Name (typically at administered with a minimum of 15 alphanumeric characters is displayed with the incoming call to the VoiceCon 08 call center.

9.6.6 Describe each of the available “states” an agent can be in, e.g., logged in; available to take calls; after call work time; etc.

Avaya Response:

To log in, an agent goes off-hook and dials the system assigned login feature access code (FAC) or depresses a pre-administered button that executes that access, followed by the agent’s assigned login ID and password, if required. If login is successful, the agent automatically enters Auxiliary Work mode for all the skills assigned to the agent and the assigned skills are displayed on the station set. The Auxiliary Work button lamp(s) on the station set lights steadily and the agent hears the confirmation tone.

At any given time, a staffed agent can be in one of four (4) unique work modes:

¾ Auxiliary (AUX) Work

¾ Auto-In

¾ Manual-In

¾ After Call Work (ACW)

An agent can change work modes at any time. To enter any work mode, an agent presses the administered and labeled button for the desired work mode or dials the Facility Access Code (FAC) for that mode, depending on what has been administered. If the agent has no active or held calls, the work-mode button lamp lights steadily and CMS or BCMS is informed of the agent’s mode change. If the staffed agent has active or held calls, the lamp flashes until all calls are dropped, then the new work mode’s lamp lights steadily and CMS or BCMS is informed of the agent’s mode change.




Auxiliary (AUX) Work mode

A staffed agent should enter Auxiliary Work mode whenever taking a temporary break. This makes the agent unavailable for ACD calls and removes them from the most-idle-agent queue. CMS and BCMS can continue to track the agent.

CMS/BCMS is notified whenever an agent in AUX Work mode receives an incoming non-ACD call or makes an outgoing call. When an agent logs into a split, he or she automatically enters AUX Work mode for that split. Agents in vector-controlled splits/skills can go into AUX Work mode even if they are the last logged-in (staffed) agent and calls are queued to that split or skill.

Auto-In mode

In an Auto-In mode, the staffed agent automatically becomes available for answering new ACD calls upon disconnecting from an ACD call.

Manual-In mode

In a Manual-In mode, the staffed agent automatically enters After Call Work (ACW) mode for the skill upon disconnecting from an ACD call and is not available for any ACD calls. To become available for ACD calls, the agent must manually reenter either auto-in mode or manual-in mode.

After Call Work mode

A staffed agent should enter After Call Work (ACW) mode when he or she needs to perform ACD-related activities, such as filling out a screen as a result of an ACD call. The agent is unavailable for ACD calls to all skills while in ACW mode. Communication server administration determines whether the agent remains in the Most Idle Agent queue while in ACW.

When an agent is in the Manual-In mode and disconnects from an ACD call, he or she automatically enters ACW mode. Although no longer available for ACD calls, the agent is still available for non-ACD calls. CMS or BCMS is notified whenever an agent in ACW mode receives an incoming non-ACD call or makes an outgoing call.

9.6.7 Describe how an agent enters work codes that describe the nature of the call. Specify the maximum number of work codes and work code digits that can be entered into the system.

Avaya Response:

Call work codes are up to 16-digit sequences that ACD agents enter to record customerrelated information. VoiceCon would define the codes for its solution. Codes that agents enter are sent to the CMS storage for skills (hunt groups) that are measured by CMS when the link to the CMS is up. Agents must have multi-appearance telephones to enter call work codes.




To enter call work codes, the agent must be off-hook and either:

¾ On an ACD call

¾ In After Call Work (ACW) mode after disconnecting from a call while in Manual-In mode remaining off-hook

¾ In Timed ACW (After Call Work) after disconnecting from a call while in auto-in mode

¾ In auto-in mode and pending for ACW mode

The sequence of event is as follows:

1. The staffed agent selects their administered Call Work Code (CWC) button.

2. The CWC lamp lights steadily and a C: prompt appears on the agent’s display. The agent must wait for the ready indication before entering the call work code or the caller hears the touch-tone digits being dialed.

3. Agent enters up to 16 digits on the dial pad. The agent can press * to erase digits.

4. The agent presses # to send the code entry to CMS.

5. The Call Work Code lamp goes dark and the display returns to normal.

6. If the agent presses any feature button or hangs up during digit collection, the code entry is cancelled and data is not sent to CMS. The CWC lamp goes dark and the display is cleared.

Call work codes may be used by as many as 100 agents simultaneously. If 100 agents are simultaneously using this function, and another agent attempts to enter a call work code, the agent receives a display message to try again later.

It is also possible to have “forced entry of stroke counts and call work codes” since you can administer a split or skill so that agents must enter a stroke count and/or a call work code before becoming available for another call using Manual-In mode.

To enter a stroke count and/or call work code, the agent must be on a call or in After Call Work (ACW) mode after releasing a call in Manual-In mode.

After releasing a call, the agent automatically enters ACW mode and cannot return to Manual-In mode until they have entered a stroke count or call work code. If the agent presses the Manual-In button or dials a Facility Access Code (FAC) before entering a stroke count or a call work code, the Manual-In lamp flutters or intercept tone is given.

The maximum number of call work codes is that which can be stored in the call work code tables on an Avaya Call Management System (CMS) is currently 1,999. This differs from the maximum number of 5,000 call work codes that can be collected in call records. It is possible to enter up to six (6) different call work codes on a single call.

9.6.8 Can agents be made automatically available immediately after each call?

Avaya Response:

Yes. The proposed ACD solution from Avaya supports the ability for staffed (logged-in) agents to be made automatically available immediately after each call.




9.6.9 Can agents be members of multiple groups/splits/skills?

Avaya Response:

Yes. The proposed ACD solution from Avaya supports the ability for agents to be members of multiple skills and in the recommended solution a single agent can be in up to a maximum of sixty (60) unique skills.

9.6.10 Can agents be made automatically unavailable after each call in order to complete work associated with the call before the next call is delivered? Can this time be specified and controlled and is this unavailable state measured and tracked in ACD reports?

Avaya Response:

Yes. The proposed ACD solution from Avaya supports:

¾ The ability for the VoiceCon 08 agents to be made automatically unavailable after each call in order to get work associated with the call [typically called either Post Call Processing – Wrap-Up – After Call Work (ACW) completed prior to the next call being delivered. There are a fair number of Avaya users who also take advantage of Timed ACW by Vector Directory Number (VDN) and by Skill (hunt group) so as to insure that all staffed agents get some administered break (perhaps two seconds) between arriving ACD calls. In the Avaya solution the Timed ACW by VDN setting overrides Timed ACW by Skill so that you can ensure you staffed agents dealing with more difficult call types maybe get a little more time between calls.

¾ This time can be specified and controlled and this “agent work state” is measure and tracked for both standard and customized ACD reporting.

9.6.11 Can agents make themselves unavailable temporarily and have this unavailable state measured and tracked in ACD reports. Can the ACD agent enter a reason code to indicate why they are unavailable and have this unavailable state measured and tracked by reason code on ACD reports?

Avaya Response:

Yes. The proposed ACD solution from Avaya supports the ability for all staffed (logged-in) agents to place themselves into Auxiliary (AUX) Work Mode and have this unavailable time measured and tracked in ACD reports.

9.6.12 Can calls that ring at an available agent’s station, but are not answered automatically, be redirected to the next available agent rather than letting the call ring unanswered until abandonment? For example, if an agent left their station without logging out, will the system automatically log the agent out or make them unavailable and notify the supervisor? Will this event be tracked by the reporting system?

Avaya Response:

Yes the proposed ACD solution from Avaya supports the ability for inbound calls that ring at an agent’s station and not automatically answered to be redirected to receive a wide range of call handling treatments rather than being allowed to ring unanswered until the caller opts to abandon.




The proposed ACD solution from Avaya does support Remote Logout of Agent feature allows a user to logout an idle agent without being physically present at the agent’s station and the user who is logging out the agent can be locally or remotely located. If an agent walks away from their station without logging out, ACD calls are sent to the station without being serviced. Without the Remote Logout of Agent feature, supervisors or other agents had to walk over to the agent’s station, enter the logout Facility Access Code (FAC) to change the agent’s work mode or log out. Customers could also busy-out the station from their server room or use the Redirection On No Answer (RONA) feature to put the station into AUX (auxiliary) work mode without logging out the agent. The Remote Logout of Agent feature makes it simpler to logout the agent from the user’s station but the system (Avaya Communication Manager) does not, at this time, actually automatically log the agent out or make them unavailable if they just leave their station without having logged out. The Supervisor is not notified but there are exception reports available from CMS.

Our ongoing customer advisory committees tell us that rather than have differing departments set up differing rules for how much time must transpire before an idle agent who is no longer present at their logged-in station would get logged-out and reported upon that the current method of relying upon CMS exception reporting and resultant human intervention appears to be preferable.

What Avaya recommends to meet this need is the Call Center feature known as Forced Agent Logout from Clock Time which allows administrators to:

¾ Set a specific time when the Communication Manager will automatically log out staffed Expert Agent Selection (EAS) agents.

¾ Define a system-wide logout reason code for agents for use with the forced logout in support of occurrence tracking from an Avaya Call Management System.

¾ Administer a forced logout override button so that staffed agents in the process on closing down as their shift and the center is closing will have the ability to override this feature, if needed.

This desirable feature would allow VoiceCon to set a pre-determined time to automatically log out staffed agents when these agents forget to log out at the end of their shifts. Using this feature assures that agents will not remain logged into the system when they are not on their shift. This allows the company to maintain accurate staffing views and prevents sending calls to auto-answer agents who have left their positions.

Avaya tracks Redirection On No Answer (RONA), Forced Agent Logout from Clock Time as well as Forced Agent Logout from After Call Work (ACW) via Reason Codes with an Avaya Call Management System (CMS).

9.6.13 Can the system provide a brief announcement heard only by the agent indicating what type of call is arriving so that the agent can greet the caller appropriately if agents handle calls for multiple applications or who are visually impaired? Can the voice terminal also display this information to the agent before delivery of the call?

Avaya Response:

Yes the proposed ACD solution from Avaya, using the Avaya S8500C Server, supports 3,000 announcements and any of these announcements can be administered as VDN of

Origin Announcements (VOA) which means that staffed agents, regardless of their visual acuity, can get announcements of up to 1.2 seconds in duration.




The proposed ACD solution from Avaya can display information to the agent prior to inbound ACD call’s delivery to the staffed agent up to the number of characters that the Agent’s telephone can support.

9.6.14 Can agents be “logical agents”, i.e., can they login with their agent ID from any system endpoint and take ACD calls?

Avaya Response:

Comply. The proposed ACD solution from Avaya, which features Call Center Software Elite, has Expert Agent Selection (EAS) active which means that all agents in the system are actually “logical” agent which have their inbound ACD calls delivered to their EAS Agent LoginID. Agents can login from anywhere in the world and have the option as part of their login process a password of up to nine (9) digits in length prior to completing their login process prior to taking calls.

9.6.15 Confirm if the system can automatically record agent calls for quality and monitoring purposes, and indicate if all calls can be recorded (incoming, station-to-station, non-ACD calls, etc.). Indicate if there is a beep tone to notify one or both of the call parties that the call is being recorded. Also indicate if the proposed ACD solution supports an integrated call recording feature or if an auxiliary system is required.

Avaya Response:

Comply. With optional recording equipment (not provided in this proposal) all calls can be recorded. The proposed ACD solution from Avaya, which features Call Center Software Elite with Expert Agent Selection (EAS) supports beep tone assignments as administered to help VoiceCon comply with state laws.

9.7 System Call Prompts & Announcements VoiceCon requires that the proposed ACD solution prompt the caller to enter digits to determine how a call should be routed and then route based upon their response.


Confirm if the proposed ACD solution has a fully integrated call prompt (call director) feature that does not require optional hardware/software equipment? If yes, identify how call prompt announcements are programmed, scripted and recorded; the maximum number of programmable call prompts; and the maximum number of concurrent activated call prompts for incoming calls. Also indicate if the caller can interrupt the call prompt for TUI response before the full script is played.

Avaya Response:

Comply. The proposed ACD solution from Avaya supports the ability using caller entered digits to determine how an arriving call should be routed and determining its ultimate response. The source of the announcements in this proposed solution are integrated. They are embedded within a media gateway processor board called a “vVAL (virtual Voice Announcement over Local Area Network) source.”

The proposed ACD solution from Avaya supports the ability to have up to 3,000 integrated announcements in software that can be deployed in support of call prompting. Call Prompting is accomplished within Call Vectoring (see Avaya Appendix 8 for examples) and the calling party can interrupt the call prompting announcement at any time as it is being played. It is possible for up to 1,000 callers to be hearing the same announcement, at the same time. The announcements are recorded via .wav files and administered via System Administration.




There are no actual limits as to the maximum number of programmable call prompts but there is a limit of 3,000 announcements within software and there are a maximum of 2,000 call vectors with 99 addressable command lines each, meaning there theoretically be up to 198,000 call prompting commands active.

Announcements can be administered to gateway and files can be FTP’ed to that gateway. The embedded VAL circuitry on each individual Gateway processor circuit board must be assigned on the Media-Gateway screen and enabled using the enable announcements command before announcements can be recorded using the telephone or played from that gateway. These commands and their description are available at www.support.avaya.com within document labeled “Maintenance Commands for Avaya Communication Manager, Media Gateways and Servers, 03-300191.” The detailed instructions for administering announcements (recording – copying – deletion, etc.) as well as troubleshooting guidelines can be found within the document named “Feature Description and Implementation for Avaya Communication Manager” at this same web site.

9.7.1 Hands Free Caller Prompt Response Vendor Response Requirement:

Confirm if the proposed ACD solution can support integrated Automatic Speech Recognition (ASR) as an alternative to TUI and describe how speech prompts are scripted and recorded.

Avaya Response:

Since the proposed ACD solution from Avaya does not include an Avaya Voice Portal because the use of Speech Recognition was not specified as an alternative to the integrated Call Prompting (the TUI) we suggest reviewing our publically available Avaya Documentation at www.support.avaya.com. From the perspective of handling inbound ACD calls, the IVR needs to map the call to a 3 to 7 digit Vector Directory Number (VDN) which would then facilitate its call handling by staffed VoiceCon agents.

9.7.2 Announcements Vendor Response Requirement:

Confirm that the proposed ACD solution has fully integrated announcement capabilities and provide answers to the following:

Avaya Response:

The proposed ACD solution from Avaya has fully integrated announcement capabilities via H.248 Media Gateway VAL source.

The proposed Avaya ACD solution features the S8500C Server and Five (5) Avaya G450 Media Gateways so that means you will have three (3) hours and 45 minutes of Announcement Time for use in supporting your 15 Agent Groups which we will equate to 15 Skills.

VAL Sources Type of VAL Source Storage Time for Announcements Playback Ports G450 Media Gateway 45 minutes 64 G350 Media Gateway 10 minutes 6




VoiceCon would set the “queue” field to y, for yes, on the Announcements/Audio Sources screen for each extension that you want to queue for Integrated Announcements. Calls that hear integrated announcements at extensions that have “queue assigned” will only queue when all of the ports on the source that contains the announcement are busy. When a port becomes available, all callers queued to hear a specific announcement up to the maximum supported by the server platform, which is 1,000, are simultaneously connected to that port to hear the announcement from the beginning of the announcement. The same queuing pool, which in this case is 4,000 integrated announcement queue slots, is used over all integrated sources. The Communication Manager controls the announcement queue length for integrated announcements, but you must set the queue length for analog or aux-trunk announcements.

Announcements are wave files that are recorded as CCITT u-law/a-law, 8kHz, 8-bit mono files using a utility such as Microsoft’s Sound Recorder on a computer or using a telephone set off the Avaya Communication Manager. Any announcement that has not been administered for barge-in use can be played through multiple ports.

Each newly available G450 Media Gateway supports, in its initial release 45 minutes of recorded announcements which might well be enough for the needs of VoiceCon in support of its 15 Agent Groups. Each G450 Media Gateway VAL source supports 64 play ports which is a significant enhancement over the G700 MG VAL source which is limited to 15 play ports and the G350 MG VAL source with just 6.

• How many different announcements can be provided?

Avaya Response:

Since the proposed ACD solution from Avaya will be featuring the Avaya S8500C Server then Avaya will be supporting the ability to have up to a maximum of 3,000 different announcements. If more announcements are required we’d suggest using the Avaya S8730 server instead since it supports up to 9,000 announcements.

• How many announcements can be played concurrently?

Avaya Response:

Avaya has proposed the Avaya G450 Media Gateway that means the 15 agent groups you wish to support will be supported with 64 announcement play ports and 45 minutes of announcement time in one single G450 Media Gateway. Since we have five (5) Avaya G450 Media Gateways in this solution we over three (3) hours of announcement time and 250 announcement play ports, with up to 3,000 different announcements played simultaneously (using announcement queuing).

• How many announcement boards are supported?

Avaya Response:

Since the proposed ACD solution from Avaya will be featuring the Avaya S8500C Server. Avaya will be supporting a maximum of 250 vVAL (virtual VAL) announcement sources on Avaya G350, G450 Media Gateways provided.

• Are digitally recorded announcements supported?

Avaya Response:

Avaya will be supporting a maximum of 3,000 announcements and/or audio (music) sources which can be digital. These are integrated through an analog port.




• How many different music sources can be supported?

Avaya Response:

Avaya can support 3,000 differing music sources in the proposed design if we were to have no announcements. Each separate music source would require an analog port.

• Can multiple announcements and music treatment be provided to a call, and can announcements and music treatment be specific to each queue?

Avaya Response:

Yes, additionally announcements and music treatment can be specific to each queue where queue equates to skill (hunt group). With 2,000 vectors, now up to 99 steps each and capable of being linked together it is technically possible to have a single call exposed to a maximum of

198,000 call control command lines.

• Can announcements and music treatment provided depend upon queue conditions or call related information, and how many different announcements can

be provided for this situation?

Avaya Response:

Yes, in the proposed ACD solution from Avaya both announcement and music-on-delay treatment can be provided depending upon either queue conditions and/or call related information. Since Avaya has proposed an S8500C Server this proposed ACD solution has a maximum of 3,000 announcement and/or audio (music) sources. If more are required, we would recommend the S8730 Server which has the ability to support a maximum of 9,000 announcement and/or audio (music) sources with up to a maximum of 128 hours of VAL board recording time.

• How do you handle feedback (music/announcements) for calls that are queued remotely?

Avaya Response:

This is accomplished in a multi-site enterprise that uses Multi-Site Best Service Routing, marketed as Avaya Virtual Routing, using a feature called “BSR Local Treatment for Calls Queued Remotely over IP or ISDN Trunks.” In a multi-site BSR configuration, an inbound call that arrives at a local communication server can be rerouted to a remote server located in a different part of the world. To better meet the needs of Avaya created this capability years ago, in Avaya Communication Manager 2.0, to allow a multi-site customer to provide local audio feedback for IP and ISDN calls while an inbound ACD call waits in queue on a remote server.

This feature provides the following potential benefits for call center operations:

¾ For multi-site BSR operations that include sites located in different countries, the new local treatment feature can result in significant bandwidth savings for IP calls.

¾ Audio quality concerns that occur when music is sent over wide area networks that use low bit-rate codecs are eliminated.

¾ Announcements and other treatments can be maintained and managed in a central location.

The local treatment operations described above assumes that the required feature and vector administration steps are implemented on both the local and remote communication servers.




• Can you connect audible feedback locally for calls that are queued remotely in order to decrease the number of packets sent over the IP trunk?

Avaya Response:

Yes, Avaya can connect audible feedback via Local Sourced Music and Announcements for remotely queued calls so as to decrease the quantity of packets sent over IP trunking.

9.8 MIS Reporting System VOICECON requires a comprehensive MIS reporting system capability with its proposed ACD solution.


Confirm that the proposed ACD MIS reporting system supports each of the following capabilities and attributes, and identify any that are not supported:

• Track local and remote, digital and IP agents

Avaya Response:

Comply, the proposed ACD Management Information System (MIS) reporting system from Avaya supports tracking of local and remote agents regardless of if the call taking agents are using digital communications processing (DCP) type telephone instruments, IP type telephones instruments, Avaya IP Agent – Shared Control or Avaya IP Agent (typically deployed at remote locations).

• Windows-based GUI

Avaya Response:

Comply, in the proposed ACD MIS reporting system from Avaya, a windows based graphical user interface (GUI) is supported in behalf of the of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Real-Time Monitoring

Avaya Response:

Comply; in the proposed ACD MIS reporting system from Avaya, real-time monitoring is supported in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Reporting Exceptions

Avaya Response:

Comply, in the proposed ACD MIS reporting system from Avaya, reporting exceptions are supported in behalf of the of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Threshold Notification

Avaya Response:

Comply; in the proposed ACD MIS reporting system from Avaya, threshold notifications are supported in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.




• Web-based access to reports

Avaya Response:

Comply, in the proposed ACD MIS reporting system from Avaya, web-based access to reports are supported in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Historical Reporting

Avaya Response:

Comply, in the proposed ACD MIS reporting system from Avaya, Historical Reporting is supported in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Custom Reporting Option

Avaya Response:

Comply, in the proposed ACD MIS reporting system from Avaya, custom reporting options in the areas of real-time, historical and integrated (combined real-time with historical) reporting are supported in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Open Database Connectivity

Avaya Response:

Comply, in the proposed ACD MIS reporting system from Avaya, open database connectivity (ODBC) is supported in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Exporting Data to other applications

Avaya Response:

Comply, in the proposed ACD MIS reporting system from Avaya, the ability to export data to other applications is supported in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Local and remote access by supervisors

Avaya Response:

Comply, in the proposed ACD MIS reporting system from Avaya, both local and remote access to reporting is supported in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

9.8.1 System Requirements Vendor Response Requirement:

Briefly describe the hardware/software requirements for the proposed ACD MIS reporting system, including requirements for applications servers and software, workstation terminals (agent and supervisor) and client software.

Avaya Response:

Avaya CMS R14 includes the Sun Netra 210 which was first introduced to the marketplace with Avaya CMS R13.1.




Sun Netra 210, initial model:

¾ One 1.34 GHz CPU

¾ 1-GB RAM

¾ One Internal 73-GB SAS disk drive; two if mirrored

¾ Four built-in Ethernet ports, auto sensing 10/100/1000 Mbps, numbered bge0 through bge3

¾ Internal DVD-RW drive

¾ External DAT 72 tape drive

Prior to beginning an installation we’d want to verify that all the following components are on hand:

¾ Computer and power cord

¾ Tape drive, power cord, and SCSI cable

¾ Monitor, cable, and monitor AC power cord

¾ USB keyboard and cable

¾ USB mouse and cable

¾ A package of blank tapes for backups

¾ One tape that contains the Avaya factory configuration CMSADM file system backup

¾ Category 5 LAN cable

¾ Modem and cables

¾ Sun and CMS software

To install and run CMS Supervisor, on VoiceCon provided desktop PC’s the system must meet the following minimum requirements:

¾ A Pentium-class or compatible processor rated at the appropriate speed:

◊ Windows 2000, XP - 300 MHz or faster

¾ 50 MB of free hard disk drive space - If you want to install CMS Supervisor in more than one language, you also need an additional 5 MB of disk space for each installation of CMS Supervisor in another language.

¾ A CD-ROM or DVD drive

¾ RAM: - Windows 2000, XP - 128 MB However, if simultaneous CMS Supervisor instances run in the background, additional memory may be required to maintain acceptable performance.

¾ An SVGA monitor and graphics adapter set to at least 64,000 colors and a resolution of at least 800x600.

¾ A network, serial, or modem connection

If you want more than one version of CMS Supervisor to run simultaneously on a PC, you will need more memory and disk resources then these absolute minimums we have stated above.




With a network connection, you would need one of the following:

¾ TCP/IP network protocol

¾ An Ethernet communications board

These requirements include Secure Shell (SSH) connections.

With a serial connection, you would need one of the following:

¾ A serial cable that is correctly wired to support hardware flow control and a 16550A UART communications port

¾ An available COM port

For a modem connection, you will need at least a 19.2 Kbps modem and a 16550A UART communications port. If the modem is external, the cable must be correctly wired to support hardware flow control.

9.8.2 MIS Reports The ACD MIS reporting system must be able to support a wide variety of report categories, including, as a minimum, the following:

• Single Agent Reports

Avaya Response:

Comply, in the proposed ACD MIS reporting system, Avaya supports Single Agent Real-Time Reports, Single Agent Historical Reports, Single Agent Integrated Reports and Single Agent Exception Reports in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Agent Group Reports

Avaya Response:

Comply, in the proposed ACD MIS reporting system, Avaya supports Agent Group Real-Time Reports, Agent Group Historical Reports, Agent Group Integrated Reports and Agent Group Exception Reports in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

• ACD Queue Reports

Avaya Response:

Comply, in the proposed ACD MIS reporting system, Avaya support ACD Queues as Skills. This means support in included for (Real-Time Skill Reports, Historical Skill Reports, Integrated Skill Reports and Skill Exception Reports in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Abandoned Calls Report

Avaya Response:

Comply; in the proposed ACD MIS reporting system, Avaya support Real-Time Abandoned Calls Reports, Historical Abandoned Calls Reports, Integrated Abandoned Calls Reports and Abandoned Calls Exception Reports in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.




• Trunk Reports

Avaya Response:

Comply; in the proposed ACD MIS reporting system, Avaya support Real-Time Trunk Reports, Single Agent Historical Reports, Single Agent Integrated Reports and Single Agent Exception Reports in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Daily Total Reports

Avaya Response:

Comply; in the proposed ACD MIS reporting system, Avaya support Daily Total Reports Single Agent Real-Time Reports, Single Agent Historical Reports, Single Agent Integrated Reports and Single Agent Exception Reports in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.

• Ongoing Status Reports

Avaya Response:

Comply, in the proposed ACD MIS reporting system, Avaya supports Ongoing Real-Time Status Reports, Ongoing Historical Status Reports, Ongoing Integrated Status Reports and Ongoing Status Exception Reports in behalf of the needs of the RFP specified 5 active and 10 configurable supervisors.


Confirm the proposed fully integrated ACD MIS reporting system satisfies this requirement, and identify any listed report type not supported.

Avaya Response:

Comply.

9.8.3 Real Time Reports Vendor Response Requirement:

Provide a list of all standard and optional real time monitoring reports.

Avaya Response:

The following table lists the Avaya CMS Supervisor reports that are available. This table recaps real-time, historical and integrated (combined real-time and historical). The reports you will see depend on your Communication Manager type, permissions, and system performance. CMS Graphical Reports are indicated in the report name.

CMS Report name Real-Time

Historical Integrated

Outbound Split/Skill

Agent Attendance

Agent AUX (Auxiliary Work)

Agent Event Count

Agent Graphical Information

Agent Graphical Time Spent

---

---

---

---

---

---

Yes

Yes

Yes

Yes

---

Yes

---

---

---

---

Yes

---


---

---

---

---

---

---

---

---

---

---

---

---

---

---

---

---

---

---

---

---





Agent Group Attendance

Agent Group AUX (Auxiliary Work)

Agent Group Report

Agent Group Summary

Agent Inbound/Outbound

Agent Information

Agent Login/Logout (Skill)

Agent Login/Logout (Split)

Agent Report

Agent Split/Skill

Agent Status by Location

Agent Summary

Agent Trace

Busy Hour by Trunk Group

Busy Hour by VDN

Call Record

Call Work Code

Event Count Summary

Graphical Active Agents

Graphical Allocated Agents

Graphical AUX Reserve1 Agents

Graphical AUX Reserve2 Agents

Graphical Average Positions Staffed

Graphical Busy/Abandon/ Disconnect

Graphical Maximum Delay

Graphical Multi-ACD Service Level Daily

Graphical Queue

Graphical Skill Overload

Graphical Split/Skill

Graphical Split/Skill Call Profile

Graphical Split/Skill View

Graphical Staffing Profile

Graphical VDN Call Profile

Multi-ACD

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

---

---

--- ---

Yes

Yes

---

Yes

Yes

---

---

Yes

---

---

--- ---

Yes ---

--- ---

Yes

Yes

Yes

Yes

Yes

Yes

---

---

---

---

---

---

--- ---

--- ---

--- ---

--- ---

--- ---

Yes

Yes

Yes

Yes

---

---

---

---

--- ---

--- Yes

--- ---

--- ---

--- Yes

--- ---

--- ---

Yes ---


---

---

---

---

---

---

---

---

---

---

---

---

---

---





Multi-ACD by Split/Skill

Multi-ACD Call Flow by VDN

Multi-ACD Top Agent

Queue/Agent Status

Queue/Agent Summary

Queue/Top Agent Status

Reserve1 AUX Agents

Reserve2 AUX Agents

Skill AUX Report

Skill Status

Skill Top Agent Report

Split Status

Split/Skill Average Speed of Answer

Split/Skill by Location

Split/Skill Call Profile

Split/Skill Comparison

Split/Skill Graphical AUX Agents

Split/Skill Graphical AUX Top Agents

Split/Skill Graphical Call Profile

Split/Skill Graphical EWT

Split/Skill Graphical Service Level

Split/Skill Graphical Status

Split/Skill Graphical Time Spent

Split/Skill Graphical Top Skill Status

Split/Skill Outbound

Split/Skill Queue

Split/Skill Report

Split/Skill Service Level

Split/Skill Status

Split/Skill Summary

System

System Multi-ACD

System Multi-ACD by Split/Skill

Top Agent Status

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

---

---

---

---

--- ---

--- ---

--- ---

--- ---

Yes

---

---

---

--- ---

--- ---

--- ---

Yes

---

---

---

Yes

---

---

---

Yes

---

--- ---

Yes

---

---

---

Yes

---

---

---

Yes

---

---

---

Yes

---

---

---

Yes

Yes

Yes

Yes

Yes

Yes

Yes

---

---

---

---

---

---

---

---

---






Trunk --- Yes ---

Trunk Group Yes Yes ---

Trunk Group Summary --- Yes ---

VDN (Vector Directory Number) Call Handling --- --- Yes

VDN Call Profile Yes Yes ---

VDN Multi-ACD Flow --- Yes ---

VDN Report Yes Yes ---

VDN Service Level --- Yes ---

VDN Skill Preference Yes Yes ---

Vector Yes Yes ---

Work State Report for Reserve1Agents Yes --- ---

Work State Report for Reserve2Agents Yes --- ---

9.8.4 Historical Reports Vendor Response Requirement:

Provide a list of standard and optional historical system reports.

Avaya Response:

Please review our answer to 9.8.3 as this table contains the list of standard historical reports, real-time reports, and integrated reports which are mix of real-time and historical reports.

9.8.4.1 Frequency Vendor Response Requirement:

What is the frequency that reports can be produced?

Avaya Response:

The Avaya CMS Supervisor reports can be generated on demand and per set timetables to meet the frequency that VoiceCon 08 requires to support its call center operations.

9.8.4.2 On-Demand Reporting Vendor Response Requirement:

Can all of the historical reports be accessed on-demand? Identify any report that cannot be accessed ondemand

Avaya Response:

Yes, all historical reports are accessible on demand.




9.8.4.3 Storage & Backup Vendor Response Requirement:

How long are historical reports stored and archived by the system, and describe data backup operations.

Avaya Response:

Avaya Call Management System (CMS) statistics for agents, split/skills, trunks, trunkgroups, vectors, and VDNs are stored in customer-defined intervals (15, 30, or 60 minutes system-wide) for up to 62 days. Daily statistics are stored in 24-hour intervals for up to five years; weekly and monthly summary data can be stored for up to ten years. Up to 2,000 customerdefined exceptions are saved; and up to 15 days of Special Days forecasting information can be stored.The raw historical interval data is stored and used for the ad hoc generation of historical reports coveringany period of time within the storage intervals. The system supports expandable storage capacities toprovide long-term data storage up to these system maximums.

Avaya CMS maintenance backups save only AvayaCMS data (administration and historical). The Avaya CMS data for each Automatic Call Distribution (ACD) which equates to Avaya Call Center Software - Elite should be backed up:

¾ After the Avaya CMS is initially provisioned

¾ After the Avaya CMS software is upgraded

¾ On a daily basis.

The CMSADM file system backup saves all local file systems on the Netra 210 onto a tape, including:

¾ Sun Solaris system files and programs

¾ Avaya CMS programs

It is strongly recommended that VoiceCon 08 not use the original set of factory backup tapes or provisioning backup tapes as that backup contains the default factory configuration. Ideally these tapes would be saved and never reused in rare case the system needs to be reinstalled in the field.

To perform a CMSADM backup:

1. Verify that:

¾ The computer (Netra 210) is in a Solaris multi-user state (2 or 3). To check whether you are in the multi-user state, enter: who –r

¾ You are using the correct tape for the tape drive on your system. VoiceCon 08 should use a new set of backup tapes for this CMSADM file system backup and avoid using the original set of factory backup tapes. Make sure you have enough tapes to execute the backup.




2. Log in as root.

3. Enter: cmsadmin and the system will display the Avaya Call Management System Administration Menu

4. Enter the number associated with the backup option. Depending upon the system configuration of your system, the system displays one of the following options:

• If only one tape drive is available on the system, go to step 5.

• If more than one tape drive is available for use by the system, the system displays a list of tape devices. Enter a tape drive selection from the displayed list.

The system displays the following message:

Please insert the first cartridge tape into <device name>.Please ENTER when ready or Del to quit:^?

5. Press Enter and the backup process begins. If more than one tape is required, the system displays the following message:

End of medium on “output”. Please remove the current tape, number it, insert tape number x, and press Enter

6. If the system displays the message in Step 5, insert the next tape and allow it to rewind. When it is properly positioned, press Enter.

7. When the backup is completed, the system displays information according to the number of tapes that are required for the backup:

¾ If the number of tapes required is one then go to Step 10 and the system displays the following message:

xxxxxxx blocks Tape Verification xxxxxxx blocks WARNING: A CMS Full Maintenance Backup in addition to this cmsadm backup must be done to have a complete backup of the system…

Please remove the current tape, number it, insert tape number x, and press Enter

¾ If the number of tapes required is more than one, the system displays the following message:

xxxxxxx blocks Tape Verification Insert the first tape Press Return to proceed:

8. Insert the first tape to be used in the backup and press Enter. Wait for the LED on the tape drive to stop blinking before you remove the tape.

9. When prompted, Repeat Step 8 for any additional tapes generated by the backup process and when the final tape is verified, the program displays the following message:

xxxxxxx blocks Tape Verification xxxxxxx blocks WARNING: A CMS Full Maintenance Backup in addition to this cmsadm backup must be done to have a complete backup of the system…

Please label the backup tape(s) with the date and the current CMS version




(Rxxxxx.x)

10. Label all tapes with the

¾ Tape number

¾ Date of backup

¾ Current version of the Avaya CMS

11. Set the tape write-protect switch to read-only and put the tapes in a safe location.

You can perform these backups within the Avaya CMS software. For more information, we would refer your administrator to “Avaya Call Management System Release 14 Administration” at www.support.avaya.com.

The Avaya Call Management System (CMS) LAN backup feature provides an alternative to the traditional method of backing up and restoring data that we just described. In the past, the only way to backup and restore data was to use a tape device located with the Avaya CMS system. LAN backup allows VoiceCon 08 to back up Avaya CMS data and system information over a local area network (LAN) to a storage manager. The storage manager is a software package that controls where, how, and when the data is stored.

The Avaya CMS LAN backup feature works two ways, depending on the type of data to be backed up.

(1) Data backup: Runs through the Informix tool ON-Bar

ON-Bar interfaces with the storage manager using the X/Open Backup Services Application Programmer’s Interface (XBSA interface). ON-Bar backs up the data in the Informix database and the system tables.

(2) System backup: Uses a storage manager to manage the backup and restore

The storage manager backs up the Avaya CMS operating-system files.

¾ The benefits of using the Avaya CMS LAN backup feature are:

¾ Automated data backups

¾ Centralized storage location for backing up multiple Avaya CMS servers

¾ Faster data backups and restores (dependent on network configuration and traffic)

The following software has been certified by Avaya for use with the LAN backup feature:

¾ Avaya CMS R14 and any associated software

¾ Tivoli Storage Manager (TSM) version 5.3.0, or version 5.2.0.

The following hardware is required to use the LAN backup feature:

¾ A customer provided storage device that is capable of interfacing with the storage manager software

¾ A network server for the storage manager server software

Note: An Avaya CMS system cannot be used as the storage manager server.

For additional information about backing up data we’d recommend reviewing the Informix Backup and Restore Guide at www.support.avaya.com.

9.8.5 Customized Reporting VoiceCon may require customized ACD MIS reports in addition to system standard/optional reports.





Confirm the proposed ACD MIS reporting system supports customized reporting capabilities and provide a brief description of how customized reports can be defined and generated.

Avaya Response:

Comply the proposed ACD MIS reporting system from Avaya supports customized reporting. There are 14 tasks you need to complete to create a custom report. If you consistently do all of these tasks for each report you create, then your reports will run properly and you will be able to create them efficiently. These 14 tasks are:

1. Defining a report's name, access, type, and option

Define the name that you use both to run the report and to access the report design if you want to change the design. Define access to whether other users can run the report and copy the report’s design to create their own custom reports. Define the type as either real-time or historical.

2. Navigating Screen Painter

Access Screen Painter so that you can design the report.

3. Copying an existing report design (Optional)

Copy an existing design so that you start with existing report headings, data fields, bars, and other report features. While this is an optional step, it saves you a lot of time and trouble by copying and modifying an existing report design. Both standard and custom report designs can be copied. Most users do not consider this step to be an option due to the time savings it affords.

4. Editing a report with blocks

Edit a report design using block moves, copies, and deletions. Editing with blocks lets you rearrange and delete sections of a report design quickly and easily. Block editing is particularly convenient when you have just copied an existing report design.

5. Entering report text

Enter text to provide headings for the data fields in the report. Entering report text should be one of the first things you do so that:

¾ The text will provide you with a skeleton layout for positioning data fields

¾ The text will help you to remember what data is supposed to go in the data fields

If you copy a report design the text of that report will be included. Using the Custom Reports subsystem, you can modify the text as required.

6. Defining fields for the Report Input window

Define fields for the report input window so that users can run the report using the parameters they choose (i.e., what split, agent, time, date, and so on, the report covers). If you copy a report design, the definition of that report’s input window will be included. You can then modify the input fields as needed.




7. Defining report fields and Defining bars in a report

Define the location of fields and bars in the report, as well as the field length, bar height/length, and the data that should be displayed in the fields/bars. Defining data consists primarily of specifying which database items supply data to the fields/bars.

Your definition of report data is not complete until you complete Task 8. If you copy a report design, the definition of that report’s data fields/bars will be included. You can then modify the data fields/bars as required.

8. Defining the rows of data for a report

Define which rows of data in a table will supply data for the fields and bars you defined in Task 7. If you copy a report design, the definition of that report’s rows of data will be included. You can then modify the selection of rows as required.

9. Defining fields to show run time/date and user inputs

Define fields on the report to show when the report was run and what items the report covers, as defined in the report input window. If you copy a report design these fields will be included and you can modify the fields as required.

10.Highlighting fields

Change brightness levels (or color if you have a color terminal) or use underlines and reverse video to emphasize or de-emphasize individual fields and text in the report.

If you copy a report design, the highlighting and other video attributes will be included. You can then modify them as required.

11.Defining stationary (no-scroll) areas

Define parts of the report that will stay in the same place in the report window even when you are scrolling up and down or right and left. You will normally define no-scroll areas for column headers, column totals, and row identifiers. If you copy a report design, no-scroll areas will be included. You can then modify the non-scroll areas as required.

12.Saving the report design

Save the design of your new custom report. So that, any work you performed will not be lost.

13. Testing the report design

Test your report immediately after designing and saving it. Testing helps eliminate wasted time in running a report whose design still has errors.

14.Running Custom Reports

Run your report.




The following standard CMS reports cannot be customized:

¾ Real-time: Multi-ACD report

¾ Real-time: Multi-ACD Top Agent report

¾ Historical: Split/Skill: Status report

¾ Historical: VDN: Status report

¾ Historical: System: Multi-ACD by Split/Skill report (daily, weekly, monthly)

¾ Historical: System: Multi-ACD Call Flow by VDN report

¾ Historical: Trunk Group: Busy Hour report

¾ Historical: VDN: Busy Hour report

To design a new custom report, you would perform the following tasks in the order indicated:

1. Access the Report Select window

¾ Select the Custom Reports option on the Main Menu and press Enter. The system displays the Custom Reports submenu:

¾ Select the Edit Reports submenu option. The system displays the Report Select window:

2. Assign a name to the report

Enter a name for your report in the Report name field. The name can have up to 20 characters, including spaces. Because the name you give your report must be unique, you may want to look at existing custom report names before entering a name for your report.

Use the following steps to list existing report names:

1. Clear the fields

2. Enter an x to select a report access—global or private

3. Enter an x to select a report type—Historical or Real-time

4. Enter an x to select a report option—Single ACD Only, ACD Group Only, or Both Single ACD and ACD Group. See the description for each report option below.

5. Press Enter

6. Select List all

7. Press Enter




The List All window displays all global or private custom report names for the specified report type.

8. Check the list to ensure that the name you want to give your report has not already been used.

If you are creating a global report, its name must be different from any name in the list. If you are creating a private report, its name must be different from names of any existing global or private reports you have created. Your private report names can be the same as another user's private report names.

9. Press Exit to return to the Report Select window. The List All window closes, and the cursor returns to the Report Select window.

10. Type the name you want for your report

3. Enter a report description

If you want, enter a description of the report in the Description field. The description can have up to 50 characters, including spaces. Your description should be detailed enough to describe the report's contents accurately. Do not use the characters within the following table:

Item Character Character Description 1 \ back slash 2 ; semicolon 3 ’ grave accent 4 ~ tilde 5 “ double quotes 6 | pipe 7 * asterisk 8 ? question mark

4. Define the user ID

Enter the user ID of the owner of the report. The default is your user ID. If you are a CMS administrator, you can enter another user's ID if you want to create a custom report for that user. Do this if you want only that user to be able to run the report. : If you want to list another user’s existing custom reports, you can enter that user’s ID and do a List All. You do not have to be an administrator to list other users’ custom reports.

5. Define access to the Report

Enter an x to select a report access option, either global or private. It is a good idea to make your reports private until they have been debugged and run successfully. This prevents the possibility of other users from running reports you have saved but not yet tested. Global access to the report gives other users the following capabilities:

¾ Ability for other users can run the report




¾ Ability for other users to be able to copy the report design when designing their own custom report. See “Copying an existing report design” more information. If you select global, your report's name must be different from the name of any other custom report — global or private. Private access to the report means that only you and the CMS administrators can run the report. No other users, except for CMS administrators, can copy or run the report design for use in their own custom reports. If you select private, your report's name must be different from the names of the following:

◊ Any other private report you have created of the same type

◊ Any user's global custom report of the same type

No other CMS user, other than a CMS administrator, can modify a report design you create, regardless of whether the report is global or private. A CMS administrator always has the ability to modify your report design, even if you make it private.

¾ Define the report as either Real-Time or Historical

Enter an x to select a report type, Real-time or Historical. When you design the report on Screen Painter, you are able to access data only for the category you select. For example, if you select Real-time, you are not able to specify historical data in the report design. Also, when you go to run the report, CMS lists the report under the appropriate submenu, Real-time or Historical. You must select the Historical option if you want to include:

¾ Exceptions data

¾ Forecast data

¾ Call work code data

¾ Agent trace data

¾ Call record data.

6. Define the report option

The Global Dictionary/ACD Groups feature provides call center supervisors with transparent views of their agents regardless of which ACD they are logged into. Multi-ACD reporting also consolidates split reporting that will aggregate data from multiple ACDs. When defining the report option, the Single ACD Only option will run only when the current ACD is set to a single ACD.

Enter an x to select one of the following options:

¾ Single ACD Only (default)

¾ ACD Group Only

¾ Both Single ACD and ACD Group

If the Global Dictionary/ACD Groups feature has not been authorized the window will not display the Report option menu.

7. Save the report name

If CMS does not add the report name because the name is not unique, use List all to see what names already exist, and go back to “Assign a name to the report.”

After you have added the report name, you can access Screen Painter to begin designing the report.




9.8.6 Terminal Report Access VoiceCon requires that ACD MIS reports must be available on terminal display and paper printout and be able to be downloaded to a PC.


Confirm the proposed ACD MIS reporting system supports this requirement.

Avaya Response:

Comply and the proposed ACD MIS reporting system from Avaya is available on terminal display and paper printout with the ability to be downloaded to a Personal Computer.

9.8.7 Scheduled and Email of Reports VoiceCon requires that ACD MIS reports must have email capability and permit the supervisor to schedule.


Confirm the proposed ACD MIS reporting system supports this requirement.

Avaya Response:

Comply and the proposed ACD MIS reporting system from Avaya supports email and permits the supervisor to schedule.

9.8.8 Report Formats VoiceCon requires that the proposed ACD MIS reporting system supports a variety of graphical and file type (EXCEL, TEXT, PDF, et al.) report formats.


Confirm the proposed ACD MIS reporting system supports a variety of graphical report formats and identify by type.

Avaya Response:

Avaya CMS Supervisor with the proposed Avaya R14.0 Call Management System supports graphical, tabular and text as well as the ability to save to HTML format. It is also possible to import to Excel.

9.9. ACD Management & Administration Vendor Response Requirement:

Provide a description of ACD management & administration capabilities. Identify in your response which of the following supervisor or administrator capabilities are supported:

• Define service levels and other thresholds

• Create call flow scripts

• Enter agent PINs and create passwords

• Limit access to data for users

• Configure all peripherals (printers, faxes etc.)

• Backup and restore the database

• Configure automatic backup and recovery of customer data




• Create individual ”views” of the call-center

Avaya Response:

User with the appropriate permissions can support the requested ACD management and administration such as:

¾ The ability to define service levels and other thresholds in both reporting and call scripts (vectors).

¾ The ability to create and modify call scripts (vectors).

¾ The ability to support VoiceCon 08 Agent personal identification numbers (PINs) which as called EAS (Expert Agent Selection) Agent Login Ids and are typically 3 to 7 digits in length within the Avaya Communication Manager’s dial plan and up to a 9-digit password can be entered for the agent’s use.

¾ The ability to limit access to data for users is controlled by setting permissions.

¾ The ability to configure all peripherals (printers, faxes etc.) is limited to those using an Avaya System Administration Terminal (SAT) or with access to Avaya Site Administration (ASA). Supervisors who have access to vectoring via CMS cannot make these types of configuration changes.

¾ The ability to perform backup and restore the database.

¾ The ability to configure automatic backup and recovery of customer data is limited to those with Avaya CMS access so this reporting piece is not available to those using an Avaya SAT/SA.

¾ The ability to administer “tenant services’ is off an Avaya SAT/SA while the ability to create individual ”views” of the call-center and restore the reporting MIS database if limited to those with Avaya CMS access.



Part 2 - System Pricing Requirements

Part 2: System Pricing

1.0 System Pricing Requirements Summary system and voice terminal pricing data will be presented to VoiceCon workshop attendees and be deemed for public use.

Detailed pricing data will remain confidential, and used to verify if the proposed system configurations satisfy RFP requirements.

Installation fee pricing data is required, and must be included in the RFP response. Indicate if the proposed installation fee is based on direct sales/service or a channel partner pricing schedule.

The proposed system price must also include a 1-year warranty to the customer. If this is a pricing option in your pricing schedule include it as part of the installation fee, and identify it as such.

Avaya Response:

Read and understood

2.0 Summary Pricing – VoiceCon IPTS solution Complete and submit the following table for your proposed IPTS solution pricing summary data in an attached file using EXCEL.

The submitted data will be made available to the general public.

Component LIST DISCOUNT All Common Equipment $144,978.84 $91,647.11

Standard Generic Software License $492,050.00 $291,027.00

Optional Software Licenses (including IP port license fees, all soft Licenses are Universal included client license fees, mobile license fees) in above

Voice Terminal Equipment $765,010.00 $505,566.50

Systems Management $0.00 $0.00

Voice Messaging $171,610.00 $113,262.60

UC System Options (includes UC client licenses) $37,973.85 $25,062.74

ACD System Options $113,069.00 $77,947.00

Installation/Warranty fees $384,052.01 $342,941.99

TOTAL $2,108,743.70 1,447,454.94

Please also see Avaya Excel file “Part 2 VoiceCon 2008 System Summary Pricing.xls”




2.2 Software Upgrade Subscriptions Complete and submit the following software upgrade subscription pricing table for your proposed IPTS solution in an attached EXCEL file.


Software Element List Discount Annual Standard Generic Software Upgrade Subscription Fee (per $4.06 $2.68 user) Annual Voice Mail Software Upgrade Subscription Fee (per user) $1.31 $0.86

Annual UC System Software Upgrade Subscription Fee (per user) $7.32 $4.83

Annual ACD System Software Upgrade Subscription Fee (per user) $9.45 $6.24

Annual Mobile Client Software Upgrade Subscription Fee (per user) Part of UC Pricing

Please also see Avaya Excel file “Part 2 VoiceCon 2008 System Summary Pricing.xls”

3.0 Desktop Voice Terminal Pricing Complete and submit the following table for your proposed unit pricing for voice terminal equipment in an attached file using EXCEL.


Voice Terminals LIST

Terminal LIST DISCOUNT Economy Desktop Telephone $166.50 $113.22

Administrative Desktop Telephone $525.00 $346.50

Professional Desktop Telephone $525.00 $346.50

Executive Desktop Telephone $679.00 $448.14

ACD Desktop Telephone $525.00 $346.50

Attendant Soft Console $2095.00 $1382.70

IP Audio Conferencing Unit $1325.00 $874.50

IP Terminal License Fee Universal License for all Ports now part of Software

Mobile Extension License Fee $115.00 $75.90

Advanced Mobile Client License Fee Included with Mobile Extension License Fee

Desktop Telephone Options Key Module Add-on $225.00 $148.50

Gigabit Ethernet Adapter $150.00 $99.00

Bluetooth adapter $95.00 $67.20

USB adapter Included as part of most 9600 Phones




4.0 Detailed Configuration Components and Pricing Submit a separate EXCEL file with a detailed listing of proposed communications system components/elements and associated unit pricing. If possible provide english language descriptions of all price configuration system components and elements in addition to any proprietary order codes.

At minimum, the configuration component list should contain:

• All common control elements (cabinet or server equipped with processor, memory, power, and interfaces)

• All common equipment port cabinets/carriers

• All port circuit interface cards for station and trunk ports

• All media gateway equipment for station and trunk ports

• All call control signaling interface cards

• All voice terminals, including audioconferencing units

• Generic software

• All port license fees

• All optional software packages

• Include all optional adjunct server equipment in support of required features and functions

• All voice messaging system elements, including cabinet equipment and memory storage

• All systems management elements

The detailed pricing will NOT be made public, but will be used to verify adherance to system configuration performance requirements and the pricing summary data.

Avaya Response:

Comply A separate detailed pricing spread sheet has been provided with the submission. Please see Excel file Part 2 Section 4 VoiceCon 2008 Proprietary detail Pricing.xls



Avaya Communication Manager, Release 5.0 Standard and Optional Features A major advantage of an Avaya solution is Avaya Communication Manager Software, which continues to set the standards for enterprise communications. The total feature-set includes in excess of 700 features. These features can help your business increase productivity, reduce costs, increase revenue, enhance customer service, boost operations between multiple locations, enhance security, increase emergency responsiveness, simplify tracking and accountability, and allow your business to customize a solution that helps meet or exceed your enterprise communications requirements.

Additional station user features which were not included in Table 9 are highlighted below in GREEN

Additional station user features which were not included in Table 10 are highlighted below in PURPLE

Additional station user features which were not included in Table 11 are highlighted below in RED

Following are of some of the key features provided by Avaya Communication Manager, Release 5.0, running on the following Avaya Media Servers: • Avaya S8730 Media Server • Avaya S8720 Media Server • Avaya S8710 Media Server • Avaya S8700 Media Server • Avaya S8500 Media Server • Avaya S8400 Media Server • Avaya S8300 Media Server

An S in the right column indicates the feature is supported via standard Avaya Communication Manager Software. In some cases, additional hardware may be required. For example, Digital Signal Level 1 (DS1) Trunk Service is a standard supported feature that requires a DS1 Interface circuit pack. An O in the right column indicates optional feature.

AAR/ARS Dialing without Feature Access Code

Feature

S

Standard (S) or Optional (O)

AAR/ARS Overlap Sending S

AAR/ARS Partitioning S

AAR Digit Conversion S

Abandoned Call Search S

Abbreviated Dialing S

Abort Transfer S

Absorb Digits S

Access Security Gateway O

Add Remove/Skills by Feature Access Code Oi



Standard (S) orFeature Optional (O)

Administrable Inter-Digit Timeout O

Administrable Language Displays S

Administrable Logins S

Administrable Loss Plan S

Administered Connections S

Administration Change Notification S

Administration Without Hardware S

Advanced Private Line Termination S

Advice of Charge O

Agent Call Handling S

Agent Call Handling Preference O i

Alphanumeric Dialing S

Alternate Facility Restriction Levels (FRLs) S

Alternate Operations Support System Alarm Number S

ANI/Information Identifier (II) Digits O

Answer Detection by Call Classifier S

Asynchronous Transfer Mode (ATM) connectivity O ii

ATM Circuit Emulation Service Licence (ATM-CES) Oii

ATM Port Network Connectivity License (ATM-PNC) O ii

ATM WAN Spare Processor License (ATM-WSP) O ii

Attendant Auto-Manual Splitting S

Attendant Auto Start and Don’t Split S

Attendant Back-up S

Attendant Call Waiting S

Attendant Calling of Inward Restricted Stations S

Attendant Control of Trunk Group Access S

Attendant Direct Extension Selection with Busy Lamp Field S

Attendant Direct Trunk Group Selection S

Attendant Display S

Attendant Intrusion (Call Offer) S

Attendant Override of Diversion Features S

Attendant Priority Queue S




Attendant Recall S

Attendant Release Loop Operation S

Attendant Room Status S

Attendant Serial Calling S

Attendant Timed Reminders and Recall S

Attendant Trunk Group/Busy Warning Indicators S

Attendant Trunk Identification S

Attendant Vectoring S

Audible Message Waiting S

AUDIX One-step Recording O

Audio Information Exchange (AUDIX®) Interface O

Authorization Codes and Authorization Codes by COR S

Auto Attendant S

Auto Available Split S

Auto Digit Rotation for Direct Inward Dial S

Auto Fallback to H.248 Media Gateways O ii

Automated Attendant (Integrated) O i

Automatic Alternate Routing (AAR) S

Automatic Callback S

Automatic Callback – Analog Telephones S

Automatic Call Distribution (ACD) S

Automatic Call Distribution Auto-Available Split (AAS) S

Automatic Call Timer with Administrable Timeout S

Automatic Circuit Assurance S

Automatic Customer Telephone Rearrangements S

Automatic Exclusion S

Automatic Hold S

Automatic Incoming Call Display S

Automatic Number Identification S

Automatic Route Selection (ARS) S

Automatic Selection of DID Numbers to Guest Rooms S

Automatic Transmission Measurement System (ATMS) S




Automatic Wakeup S

Backup and Restore S

Basic Call Management System (BCMS) S

BCMS/VuStats Login IDs S

Best Service Routing™ (BSR) O i

BSR Polling over IP without Bearer Channel O i

Bridged Call Appearance – Multi-Appearance Voice Terminal S

Bridged Call Appearance – Single-Line Voice Terminal S

Busy Verification of Terminals and Trunks S

Button View S

Call Admission Control: Bandwidth Management S

Call-By-Call Service Selection S

Call Center Basic for up to 5200 Basic Agents S

Call Coverage S

Call Detail Recording (CDR) S

Call Detail Recording (CDR) Account Code Dialing S

Call Detail Recording (CDR) Display of Physical Extension S

Call Detail Recording (CDR) Privacy S

Call Detail Recording (CDR) Record Splitting S

Call Preservation during Software Updates for Linux-based Servers S

Caller ID (Analog) S

Caller Information Forwarding (CINFO) S

Call Forwarding All Calls S

Call Forward Busy/Don’t Answer S

Call Forwarding Override S

Calling Party Number/Billing Number Restriction S

Call Management System (CMS) Oiii

Call Park S

Call Pickup (Group and Directed) S

Call Pickup Alerting S

Call Pickup of Intercom Calls S

Call Progress Tones (administrable frequencies and cadences) S




Call Prompting O i

Call Recording via Communication Manager API O

Call Vectoring O i

CallVisor® Adjunct/Switch Application Interface (ASAI) O

Call Waiting Termination S

Call Work Codes O i

CAMA Trunk Service S

Centralized Attendant Service (CAS) O ii

Centralized Voice Mail via Interswitch Mode Code S

Check In/Check Out S

Circular Hunting S

CLAN Load Balancing S

Class of Restriction (COR) S

Class of Service (COS) S

Code Calling Access S

Conference – Attendant S

Conference – Terminal S

Conferencing – Selective Display & Drop, Toggle/Swap, and Display Prompts, Selective Conference Far End Party Mute S

Conferencing – Meet Me S

Connection Preservation failover/failback for H.248 Media Gateways O ii

Connection Preserving Upgrades for Duplex Servers Oiv

Consult S

Coverage Answer Group S

Coverage of Calls Redirected Off-Net S

Coverage Callback S

Coverage Incoming Call Identification (ICI) S

Crisis Alert to Attendant, Extension, and Pager S

Custom Selection of VIP DID Number for Guest S

Customer-Provided Equipment (CPE) Alarm S

Customer Telephone Administration S

Daily Wakeup S




Data Call Setup S

Data Hot Line S

Data-Only Off Premises Extensions S

Data Privacy S

Data Restriction S

DCS (Distributed Communications System Networking Software) O ii

DCS Alphanumeric Display for Terminals O ii

DCS Attendant Control of Trunk Group Access O ii

DCS Attendant Direct Trunk Group Selection O ii

DCS Attendant Display O ii

DCS Automatic Callback O ii

DCS Automatic Circuit Assurance (ACA) O ii

DCS Busy Verification of Terminals and Trunks O ii

DCS Call Coverage O ii

DCS Call Forwarding All Calls O ii

DCS Call Waiting O ii

DCS Distinctive Ringing O ii

DCS Enhanced O ii

DCS Leave Word Calling O ii

DCS Multi-Appearance Conference/Transfer O ii

DCS Over ISDN-PRI D-Channel (DCS+) O ii

DCS Over Analog Trunks via PPP O ii

DCS with Rerouting O ii

DCS Trunk Group Busy/Warning Indication O ii

Default Dialing S

Dial Access to Attendant S

Dial-by-Name (Integrated Directory) S

Dialed Number Identification Service (DNIS) S

Dial Plan (3, 4, 5, 6, or 7 digits) S




Dial Plan: Access to local resources S

Dial Plan: Duplicate extensions per location S

Dial Plan: Local dial access code S

Digital Multiplexed Interface S

Digital Signal Level 1 (DS1) Trunk Service S

Direct Department Calling (Linear Hunting) S

Directed Call Pickup S

Direct Inward and Outward Dialing (DIOD) – International S

Direct Inward Dialing (DID) S

Direct Outward Dialing (DOD) S

Display VDN for Route-to Direct Agent Call O

Distinctive Ringing S

Do Not Disturb S

DTMF over H.245 in band S

Dual Wakeup S

Duplication of Processor Complex and Port Network connectivity O v

Echo Cancellation (integrated in DS1/E1/ISDN interface circuit packs) S

E1 Trunk Service S

E911: Call forward returned E911 dropped calls S

E911: Device Location for IP Telephones (layer 2) S

E911: ELIN by range of IP Addresses S

E911: Emergency Calls from Un-named IP Phones (Release 3.0.1) S

E911: Integration with RedSky Technology Products Ovi

E911: Use the physical set’s extension fro bridged 911 calls S

EIA Interface S

Emergency Access to the Attendant S

End-to-End Signaling S

Enhanced 911 CAMA Service Access O

Enhanced Voice Terminal Display S

Enterprise Survivable Server (ESS) O ii

Estimated Wait Time Routing O 1




Expanded Meet Me Conferencing (up to 300 users/call) Ovii

Expert Agent Selection O i

Expert Agent Selection PHD (each agent with up to 20 skills and 16 skill levels) O i

Extended Trunk Access S

Extended User Administration of Redirected Calls S

Extension Number Portability S

Extension to Cellular (EC500) Oviii

External Device Alarming Interface S

Facility and Non-Facility Associated Signaling S

Facility Busy Indication S

Facility Restriction Levels (FRLs) S

Facility Test Calls (with Security Measures) S

Flexible Billing O

Forced Entry of Account Codes S

Foreign Exchange Access S

Go to Cover S

Group Call Pickup S

Group Listen S

Group Paging S

H.245 Link Encryption S

H.320 Calls S

H.323 Trunks and Stations S

H.323 Link Bounce Resiliency for IP Phones S

Handsfree Answer S

Hold S

Hold – Automatic S

Holiday Vectoring S

Hot Line Service S

Hunting S

Inbound Call Management S

Incoming Call Line Identification (ICLID) Access S

Individual Attendant Access S




Information Forwarding S

Insert Digits S

Integrated Directory S

Integrated Management Standard or Enterprise Management package (depending on number of End User licenses) S

Integrated Services Digital Network—Basic Rate Interface S

Integrated Services Digital Network—Primary Rate Interface S

Intercept Treatment S

Intercom – Automatic S

Intercom Automatic Answer (Handsfree) S

Intercom – Dial S

Internal Automatic Answer S

International Operator Access – International S

Inter-Gateway Alternate Routing (IGAR) S

Inter-PBX Attendant Service S

Intraflow and Interflow S

Installation Wizard S

Invalid Number Recorded Announcement S

IP Network Regions S

IP Phone Registration & Signaling Encryption (Release 3.0.1) S

IP Quality of Service for Voice over IP S

IP Trunk and Station Access S

Last Number Dialed S

Least Occupied Agent O i

Leave Word Calling S

Leave Word Calling – QSIG/DCS O ii

Linear Hunting S

Linear and Preferential Trunk Hunting S

Line Lockout S

Link Recovery S

Listed Directory Numbers S

Locally Sourced Announcements and Music S




Local Survivable Processor licenses for Avaya S8300 Media Servers running in LSP mode O

ii

Login ID S

Long Hold Recall Warning S

Look-Ahead Interflow O

Loudspeaker Paging Access S

Loudspeaker Paging Access – Deluxe S

Low Bit Rate Modem Relay over IP S

Malicious Call Trace S

Manual Message Waiting S

Manual Originating Line Service S

Manual Signaling S

Media Encryption (Avaya Encryption Algorithm) S

Media Encryption (Advanced Encryption Standard – AES) S

Meet Me Paging S

Misoperation Handling S

Mixed Length Dial Feature Access Codes and Trunk Access Codes S

Modem over IP S

Modem Pooling S

Most Idle Agent (MIA) hunting S

Most Idle Agent Across Splits/Skills S

Move Agent/Change Splits/Skills while Staffed S

Move Agents from CMS O iii

Multi-Appearance Preselection and Preference S

Multi-Language Displays S

Multi-Level Precedence and Preemption (MLPP) developed for Government Applications O

Multimedia Call Handling O

Multimedia Class of Service/Class of Restriction O

Multimedia Hunting O

Multimedia – One Number Multimedia Complex O

Multimedia – Second Call Delivery O

Multinational Gateway Support O ii




Multiple Call Handling – Forced S

Multiple Call Handling on Request S

Multiple Locations based on Network Region S

Multiple Music Sources S

Multiple Network Regions per CLAN S

Multiquest® Flexible Billing O

Music-on-Hold Access S

Names Registration S

Native Configuration Manager S

Network Address Translation (NAT) with shuffling S

Netstat ARP (IP Serviceability Feature) S

Network Access – Private S

Network Access – Public S

Network Call Redirection S

Night Service – Hunt Group S

Night Service – Night Console Service S

Night Service – Night Station Service S

Night Service – Trunk Answer from Any Station S

Night Service – Trunk Group S

Non-DID Calling via UDP S

Off-Premises Station S

Per Button Ring Control (Ring, No Ring, Abbreviated Ring, Delayed Ring) S

Personal Central Office Line (PCOL) S

Personalized Ringing S

Personal Station Access S

Phantom Extensions S

Power Failure Transfer S

Priority Calling S

Privacy – Attendant Lockout S

Privacy – Manual Exclusion S

Private Networking S

Property Management System (PMS) Interface Access S




Pull Transfer S

QoS for IP Call Control S

QSIG Basic S

QSIG Centralized Attendant Service O ii

QSIG Inter-working with DCS O ii

QSIG Networking O ii

QSIG Rerouting O ii

QSIG-Supplementary Services O ii

QSIG Support for Unicode O ii

QSIG transfer into Voice Mail O ii

QSIG Value-Added O ii

Queue Status Indications S

R2 MFC Signaling S

Radio Paging Access S

Reason Codes O i

Recall Signaling S

Recent Change History S

Recorded Announcement Access S

Recorded Telephone Dictation Access S

Redirection On No Answer (RONA) S

Redirection On No Answer to Vector Directory Number (VDN) O i

Refresh Route Table (IP Serviceability Feature) S

Remote Access with Security Measures S

Remote Logout of Agent S

Remote Telephone Software Upgrades (4600 Series IP Telephones) S

Report Scheduler and System Printer S

Reroute of Denied Calls S

Reset Shift Call S

Resource Reservation Protocol (RSVP) Support S

Restrict Call Forward Off Net (by Class of Restriction) S




Restricted Presentation S

Restricted/Unrestricted Call Lists S

Restriction – Controlled S

Restriction – Fully Restricted Service S

Restriction – Miscellaneous Terminal S

Restriction – Miscellaneous Trunk S

Restriction – Toll S

Restriction – Voice Terminal – Inward S

Restriction – Voice Terminal – Manual Terminating Line S

Restriction – Voice Terminal – Origination S

Restriction – Voice Terminal – Outward S

Restriction – Voice Terminal – Public S

Restriction – Voice Terminal – Termination S

Ringback Queuing S

Ringer Cutoff S

Ringing – Abbreviated and Delayed S

Rotary Dialing S

Security Violation Notification (SVN) S

Send All Calls (SAC) S

Senderized Operation S

Separation of Bearer and Signaling (SBS) O ii

Service Levels S

Service Observing S

Service Observing by Class of Restriction S

Service Observing on VDNs O 1

Service Observing Remote S

Single-Digit Dialing and Mixed Station Numbering S

SIP: Instant Messaging & Presence via Converged Communications Server O

SIP Trunking O

SNMP Native Agent on Linux-based Servers S

Special Information Tones (SIT) Treatment S

Speed Dialing S




Standard Local Survivability (SLS) on the G250 Media Gateways Six

Station Hunt Before Coverage S

Station Hunting S

Station Lock S

Station Security Codes S

Station Self Display S

Station Used as Virtual Extensions S

Straightforward Outward Completion S

Stroke Counts S

Subnet Trunking S

Suite Check In S

Switch Based Bulletin Board S

System Measurements S

System Status Report S

T.38 Fax Interoperability S

Telecommuter Access port S

Temporary Bridged Appearance S

Tenant Partitioning S

Ten Digit to Seven Digit Conversion S

Terminal Self Administration S

Terminal Translation Initialization (TTI) S

Terminating Extension Group (TEG) S

Through Dialing S

Timed After Call Work (Agent pause between calls) S

Timed Automatic Disconnect for Outgoing Trunk Calls S

Time of Day (TOD) Routing S

Time of Day Clock Sync to LAN S

Touch-Tone Dialing S

Tone Detection S

Tone Generation S

Transfer S

Transfer Call Back S




Transfer – Outgoing Trunk to Outgoing Trunk S

Traveling Class Marks (requires trunk group administered as Tandem) S

Tripwire Security for Linux-based Servers S

Trunk Flash S

Trunk-to-Trunk Transfer S

TTY over IP S

Unicode Support S

Uniform Call Distribution S

Uniform Dial Plan (UDP) S

Universal Access Phone Status S

Universal Call ID S

Universal User Licenses S

Vector Directory Number of Origin Announcement (VOA) O 1

Vector Directory Number Return Destination O 1

VIP Wakeup S

Visually Impaired Attendant Service (VIAS) S

Voice Announcements Over LAN O

Voice Announcement Over LAN Manager O

Voice Messaging System Interface S

Voice Over Internet Protocol (standard with IP600 server) S

Voice Response Integration S

Voice Response Unit DTMF Feedback Signals O

Voice Terminal Alerting Options S

Voice Terminal Display S

VoIP Camp-on Busy Out S

VuStats S

Wakeup S

Web Pages for Configuration S

Whisper Page S

Wideband Switching S



i Included when Avaya Call Center Elite Software is purchased ii Standard with Avaya Communication Manager – Enterprise Edition iii Avaya Call Management System (CMS) is optionally available as an adjunct, the Move Agents from CMS feature is available

only if CMS is purchased iv Applies to duplex (S8700 or S8710) Media Servers only

v Duplication of the Processor Complex, and duplication of Port Network connectivity, are currently optionally available for

Avaya S8710 and S8700 Media Servers vi Requires optional RedSky application

vii Requires optional licensing and server

viii 5 Extension to Cellular licenses are included as standard with Communication Manager, Release 3.0 ix Requires Avaya G250 Media Gateway