3com Ipt Rel 7

http://www.3com.com06/07

Convergence Applications Suite

RFP Template AnswersRelease 7.1

June 8, 2007Version 1

IP Telephony Request for Proposal Template Answers

3Com Corporation, 350 Campus Drive, Marlborough MA 01752-3064

Copyright © 2007, 3Com Corporation. All rights reserved. No part of this documentation may be reproduced in any form or by any means or used to make any derivative work (such as translation, transformation, or adaptation) without written permission from 3Com Corporation.

3Com Corporation reserves the right to revise this documentation and to make changes in content from time to time without obligation on the part of 3Com Corporation to provide notification of such revision or change.

3Com Corporation provides this documentation without warranty, term, or condition of any kind, either implied or expressed, including, but not limited to, the implied warranties, terms, or conditions of merchantability, satisfactory quality, and fitness for a particular purpose. 3Com may make improvements or changes in the product(s) and/or the program(s) described in this documentation at any time.

3Com and the 3Com logo are registered trademarks of 3Com Corporation. VCX is a trademark of 3Com Corporation. All other company and product names may be trademarks of the respective companies with which they are associated.

3Com Convergence Applications Suite Release 7.2 Page 2


Table Of Contents

1 General Information and Instructions...............................................................................51.1 Company Profile......................................................................................................51.2 Purpose of Project...................................................................................................51.3 Goals and Objectives...............................................................................................61.4 Current Telephony Environment..............................................................................61.5 Current Data Environment.......................................................................................61.6 RFP Coordinator......................................................................................................61.7 RFP Schedule..........................................................................................................71.8 Proposal Submission...............................................................................................71.9 Vendor Site Tours....................................................................................................71.10 Proposal Questions..................................................................................................71.11 RFP Evaluation Factors...........................................................................................8

2 Requirements...................................................................................................................92.1 System Capacities...................................................................................................92.2 System Architecture...............................................................................................102.3 Redundancy and Reliability...................................................................................252.4 Scalability...............................................................................................................282.5 Interoperability.......................................................................................................302.6 Data Network.........................................................................................................332.7 IP Phones..............................................................................................................362.8 System Features....................................................................................................442.9 Calling Features.....................................................................................................482.10 Unified Messaging.................................................................................................882.11 Conferencing Requirements................................................................................1092.12 Presence..............................................................................................................1202.13 Call Center...........................................................................................................1212.14 System Administration.........................................................................................1222.15 Reporting.............................................................................................................1272.16 System Management...........................................................................................1402.17 Implementation....................................................................................................1442.18 Maintenance........................................................................................................1462.19 Training................................................................................................................147

3 Vendor Overview..........................................................................................................1484 Pricing...........................................................................................................................151



Instructions to Preparer

The purpose of this document is to provide an example RFP template that increases the opportunities for success with a winning proposal for a System i IP Telephony Suite solution.

This RFP template is intended for use by IBM Business Partners with IP Telephony VAE certification, 3Com voice-approved resellers, and 3Com sales personnel.

Follow these steps when using this RFP template:

Copy this file to another file with an appropriate file name Open the new file Change the Properties of the Word document as appropriate Change the cover page as appropriate Print this page and delete it from the file you send to the customer The customer or you must fill-in section 1 of the RFP template to identify the

customer needs, existing infrastructure, and RFP instructions If needed, delete some questions or entire sub-sections if they are not applicable to a

particular customer Send the document to the customer, where they can modify as appropriate Have the customer save the RFP in pdf format Use the RFP Q&A set of documents to respond to the RFP when it is sent out by the

customer



1 General Information and Instructions

This Request for Proposal (RFP) is being used to obtain proposals for a replacement of the current telephony systems in use at <CustomerName>’s corporate headquarters in <CorporateHQLocation> and at <NumberOfRemoteOffices> of our remote offices, including <RemoteOfficesLocations>. These locations are currently served by <LegacyPBXAge> year old equipment that is different at each location, including <IdentifyPBXs>.

This RFP is organized into the following sections:

Section 1 – General Information and Instructions Section 2 – Requirements Section 3 - Pricing

All vendors responding to this RFP must respond to section 2 and section 3 using the information provided in section 1.

1.1 Company Profile

Identify the company requesting the proposals. Include information about number of employees, business description, locations, history, and expected growth.

1.2 Purpose of Project

The purpose of this project is to replace the different telephony systems at our corporate office and remote offices with the best solution that provides a reliable, secure, scalable, and enterprise-wide communications platform designed to meet our corporate needs today and into the future. The solution must support, among other things, distributed architecture, centralized administration, inter-site dialing and directory, advanced PBX features, inter-site voice and fax mail with unified messaging, inbound call center functionality, in-house conferencing for audio and video, user-aware presence, instant messaging, user mobility, site survivability, emergency services, and support for remote and home office workers.

The project is to be implemented in phases, starting with the replacement of the telephone system at our corporate office, cutting users over to the new system in a phased approach. The next phase of the project will be to replace the telephony systems at each of our remote offices, one at a time, adding them to the new communications solution. Some of the remote offices require different levels of survivability in cases of WAN failure.

Modify this section appropriately to describe the purpose of the project and its implementation.



1.3 Goals and Objectives

The goal of this project is to replace our existing telephony systems with a state of the art communications solution with these objectives:

Comprehensive solution from industry leading manufacturer and vendors Reliable and scalable platform with minimum number of servers Cost effective total solution including product, installation, and maintenance Centralized administration and management of hardware and software Easy to use phones, applications, and features with superior voice quality Transparent features and dialing between phones regardless of location Mobility of users regardless of location Centralized auto attendants and voice messaging to all users Centralized scheduled, ad-hoc, and dial-out conferencing services for internal and

external users Choice of regular voice mail or unified messaging for all users Standards-based integration with other telephony applications Interoperability with existing PBX for migration Integration with corporate email and instant messaging applications Ability to provide integration with corporate business applications

Modify this section appropriately to describe the goals and objectives that are most important to your company.

1.4 Current Telephony Environment

Describe the current telephony environment at each location, including PBX type, physical interfaces to PSTN (i.e. analog or digital), signaling (i.e. analog, CAS, PRI, QSIG), number of lines or trunks, number of phones, and phone types. Describe how the telephone system is used by your company (i.e. incoming call handling, types of calls made by users, and outgoing call handling).

1.5 Current Data Environment

Describe the current data environment at each location, including LAN switch models, WAN router models, WAN bandwidth, and interface types. Also describe current server environment, email system, and any other data applications that are used for communications.

1.6 RFP Coordinator

Upon release of this RFP, all communications concerning the proposal must be directed to the RFP Coordinator listed below.

Name: Address:



Phone: Fax: Email:

1.7 RFP Schedule

The schedule for this project is as follows:

RFP Issued: <Day 1> Vendor Questions Due: <Day 1 + 7> Response to Vendor Questions: <Day 1 + 14> Proposals Due: <Day 1 + 21> Finalist presentations: <Day 1 + 35> Reference Checks: <Day 1 + 42> Selection of vendor and equipment: <Day 1 + 49> Implementation schedule due: <Cut-over Date - 14> Cut-over: <Cut-over Date>

<CustomerName> reserves the right to adjust this schedule as necessary.

1.8 Proposal Submission

All proposals must be received in its entirety no later than <ProposalsDueDate>. Proposal responses are preferred in electronic form as an email message attachment in PDF format providing the entire response is included in one attachment.

Proposal responses must be in the same structure as this RFP prefaced with an executive overview, requirements compliance information from section 2, and pricing information from section 3.

1.9 Vendor Site Tours

There will not be a formal proposal conference conducted for vendors wanting a site survey of the main office facilities. A site visit is not required for the proposal but <CustomerName> will be available to allow proposing vendors an opportunity to obtain first-hand exposure to the implementation environment. To schedule a site tour, contact the RFP Administrator.

1.10 Proposal Questions

RFP questions must be forwarded to the RFP Coordinator. The preferred method of questions is via email. All official questions and answers will be in writing and made available to all vendors.



1.11 RFP Evaluation Factors

<CustomerName> will evaluate the proposals to determine the most advantageous proposal. We will use the following factors to evaluate the proposals listed in order from most to least important:

Ability of the proposed system(s) to meet the stated requirements Proposed vendor experience and qualifications related to delivering, installing and

maintaining the proposed system Total cost of ownership for the proposed system References of comparable installations noting quality of past performances Documented installation plans for off hours implementation Documented training plans for users and Information Technology staff System warranty, technical support and annual maintenance offerings RFP response document completeness



2 Requirements

Vendors must provide brief, clear, and concise responses to the following requirements with illustrations where appropriate.

2.1 System Capacities

The following table describes the number of locations, expected growth, and the level of survivability required.

Location Type 3-year growth %

Feature Survivability?

PSTN Survivability?

The following table describes the number of IP phone sets that are required at each location.

Location BasicPhones

Business Phones

Manager Phones

AttendantConsoles

WiFi Phones

The following table describes the number of other types of stations that are required at each location.

Location Soft Phones with Companion IP Phone

Standalone Soft Phones

Software Attendant Console Positions

The following table describes the number of trunks and lines that required at each location.

Location DigitalTrunks

Analog Lines

PFTS Lines



2.2 System Architecture

Identify the manufacturer, make and model of the proposed solution, including a brief overview of the proposed solution.

This proposal presents a reliable, scalable, and forward-thinking solution for a centralized communications system for the main office location, branch offices, remote sites, and remote workers based on the 3Com VCX IP Telephony release 7.1 product.

3Com’s IP Telephony solution provides a standards-based pure IP converged application solution for enterprises with one or more locations requiring a highly resilient communications system, while truly lowering operational costs through a centralized management and provisioning system.

The 3Com IP Telephony architecture provides flexibility with built-in redundancy that allows you to deploy an IP Telephony solution that is highly available and scalable. As illustrated below, the 3Com IP Telephony architecture consists of an access tier and an application tier that communicate via SIP signaling. These tiers are encapsulated by administration and management functions providing connectivity, call processing, and applications that can be configured to meet your needs. 3Com has a complete set of media gateways for reliable and scalable connections to the PSTN and analog devices. In addition, 3Com offers a robust portfolio of IP phones.

The 3Com architecture transforms IP Telephony into an enterprise application by converging voice and data applications on a secure network with a common infrastructure for authentication, call control, presence, privacy, and management. This allows you to deliver centralized voice and data applications to your users regardless of their location in the network. By supporting end-to-end Session Initiation Protocol (SIP) signaling throughout the architecture, the 3Com IP Telephony solution provides scalability in both number of


FXO and FXS AnalogMedia Gateways

Digital Media Gateways

Telephony Applications

Media Gateways & SIP Devices

Administration& Management

Enterprise Management Suite,Web Provisioning,& CDR Reporting

IP Contact Center

IP Contact Center

IPMessaging

IPMessaging

IP Presence

IP Presence

IPConferencing

IPConferencing

IP TelephonyIP Telephony

PSTNPSTN

SIP

SIP IP & Wi-Fi Phones Soft Phone

AnalogPhone and Fax

InternetInternetIP

Tele Commuter


users and types of applications, including messaging, audio conferencing, video conferencing, data collaboration, instant messaging, and more. Provide a network diagram that shows the system elements of the proposed solution at each location, including a brief description of the system elements.

The following diagram illustrates the high-level architecture of the proposed solution. This is a pure IP-based solution that is based on network infrastructure that supports standards-based quality of service (802.1Q/p) with Power over Ethernet (802.3af) capability.

The key components of the proposed solution include:

Redundant set of x-series servers running IP Telephony Redundant set of x-series servers running IP Messaging IP Phones (and convergence center clients) Media Gateways for connectivity to PSTN and analog devices Centralized administration, centralized management, and Call Reporting Advanced features with phone and web-based user programmability of features

3Com has been shipping IP Telephony solutions to the small to mid-size market since 1998 and has over 30,000 IP Telephony installations in over 30 countries worldwide. With 3Com IP Telephony, you can benefit from one integrated, secure, and reliable communications solution delivered by an industry leader.

This proposal for a PBX replacement based on 3Com IP Telephony provides an excellent solution for today with the ability to scale both in size, resources, applications, remote locations, and devices in a standards-based manner as you grow.


PSTN

VCX IP Telephonyand IP Messaging

(Primary)

Analog FXOMedia Gateway

DigitalMedia Gateway

POTSPRI

Analog FXSMedia

Gateway

Analog Phones

Enterprise IP Network


(Secondary)

WAN

LAN

HQ (Regional Office)

Branch Offices

PRI

POTS

IP PhonesAdmin & WrbProvisioning,

CDR Reporting, CAS EMS and TTS

V6000

Dig GW

Fax Machines

IntelligentMirroring

Replication

Replicationfor IP Telephony

(to Primary)

FXS GWPSTNPSTN


(Primary)

Analog FXOMedia Gateway

DigitalMedia Gateway

POTSPRI

Analog FXSMedia

Gateway

Analog Phones

Enterprise IP Network


(Secondary)

WANWAN

LANLAN

HQ (Regional Office)

Branch Offices

PRI

POTS

IP PhonesAdmin & WrbProvisioning,

CDR Reporting, CAS EMS and TTS

V6000

Dig GW

Fax Machines

IntelligentMirroring

Replication

Replicationfor IP Telephony

(to Primary)

FXS GW


The proposed solution must support end-to-end SIP signaling. Describe how SIP provides end-to-end signaling in your communications architecture.

Converged networks based on Session Initiation Protocol (SIP) offer organizations efficient and cost-effective ways to communicate and share information. They enable location independence that helps mobile workers and supports the needs of businesses with geographically distributed offices. They provide easy scalability for evolving organizations. And the intelligence built into the design of SIP-based network components helps organizations achieve business continuity should unexpected events occur.

Session Initiation Protocol (SIP) – the only protocol based on nonproprietary Internet standards – is poised to become the leading option for enterprises that want to enable multiple applications, such as email, messaging, conferencing, video streaming and mobility into existing telephony applications.

Session Initiation Protocol is an extremely flexible call signaling protocol, primarily because its use is well defined. SIP is used to establish sessions between endpoints. It does not care what type of endpoints they are, nor what type of payload they are carrying. One of the most popular SIP endpoints in use today is the Microsoft XBOX (which uses SIP to connect to the Microsoft Live Communication Server for online gaming).

The VCX is an end-to-end SIP-based platform. Between the endpoints, using Session Description Protocol (SDP), an RTP (audio) stream is established between the IP phones. In this scenario, both the call signaling path and the voice path traverse the IP network exclusively. Because 3Com VCX is a pure SIP solution with adherence to the IETF standards, we are able to support a wide variety of SIP endpoints, both phone and non-phone.

Basic Characteristics of SIP:

Simple: SIP is based on a mechanism that simply initiates, terminates and modifies sessions over IP networks. These sessions could be as basic as a telephone call or as complex as a multi-party mixed media session.

Open: SIP is the mechanism that integrates services across platforms. It helps realize the true potential of IP telephony by delivering interoperability between vendors. SIP provides global connectivity without the need for central servers, and regardless of the individual service provider.

Secure: SIP’s simplicity makes it easier to interoperate within a multi-vendor network, but also more vulnerable to security abuses. This issue can be effectively resolved by building SIP capability into the enterprise firewall, strengthening authentication and encryption of messages.

Application-Rich: SIP enables users to embrace emerging IP applications: instant messaging, desktop call management, personal mobility, conferencing and many others.

Proven: MSN, AOL and Yahoo have all announced support for the SIP protocol, and more and more businesses are adopting SIP for person-to-person communications services. Microsoft has recently announced inclusion of a SIP-based communications client with the Windows operating system.



The proposed solution must support the ability to integrate additional SIP-based applications with the base system. Describe how the proposed solution provides the ability to add and integrate SIP-based applications to the base system.

As shown in the figure below, 3Com’s choice of SIP helps bring together multiple solution components within 3Com’s portfolio as well as to use SIP to interoperate with other networks and standards.

3Com has adopted a layered architecture for its Convergence Application. For example, IP Messaging, IP Telephony, IP Conferencing all run on a common server platform. By building the solution based on a layered architecture of solution components networked with each other using the SIP protocol, 3Com is able to reuse the components for multiple applications, resulting in significantly lower total cost of ownership.

One of the significant aspects of SIP is its use of common Internet services and protocols such as:

Domain Name System (DNS) and URLs for naming HTTP/text for message formatting Multipurpose Internet Mail Extensions (MIME) for application integration

SIP is also an HTTP like protocol with readable text encoding, making it relatively easy for 3rd party developers to create web applications and integrate web applications with telephony applications.

The 3Com Convergence Applications is architected to employ common session control, authentication, privacy, presence, and location services across a suite of IP Telephony, IP Messaging, IP Conferencing, and IP Contact Center applications and includes the 3Com Convergence Center client for total user control of their communications environment.


Applications

VoiceEnd Points

AnalogPhones

IP Phones

Digital Phones

Video

Station Gateways

PCs

AnalogPhones

IP Phones

PCs

Digital Phones

Video

Station Gateways

NetworksPCs

Mobility

Mail

Voice

Contact Center

Conferencing

UnifiedMessaging

Session Initiation Protocol

TDM

H.323

MGCP

PSTN

Private Networks

Network Gateways

ApplicationsApplications

VoiceEnd PointsEnd Points

AnalogPhones

IP Phones

Digital Phones

Video

Station Gateways

PCs

AnalogPhones

IP Phones

PCs

Digital Phones

Video

Station Gateways

NetworksNetworksPCs

Mobility

Mail

Voice

Contact Center

Conferencing

UnifiedMessaging

Session Initiation Protocol

TDM

H.323

MGCP

PSTN

Private Networks

Network Gateways


Identify the servers used with the proposed solution.

3Com offers two Linux-based IBM platforms: the IBM xSeries 306m and the IBM xSeries 346. The 306 includes Intel Pentium 4 processors with 800MHz front-side bus, 1MB L2 cache, Ultra 320 drives or Serial ATA with dual integrated 10/100/1000 Ethernet.

The 346 includes a powerful 3.2 GHz Intel Xeon processor, 1MB L2 cache, 2GB of 133 MHz memory, five PCI slots, eight drive bays, one serial port, three USB ports, and two 10/100/1000 Ethernet controllers.

The 346 will soon be replaced by the IBM xSeries 3650.

IBM xSeries 306m server:

Rack-mount 1U 1X Intel® Pentium® 4 processor 3.2 GHz/800 MHz, 2 MB L2 Cache Processor 2 GB Memory 160 GB 7200-RPM Serial ATA Hard Disk Drive On-board 10/100/1000 Ethernet 3.5” 1.44 MB Diskette Drive 24X EIDE Internal CD-ROM 1 x Serial Port 4 x USB Ports 2 x Ethernet Ports 1 x each of: Keyboard, video and mouse ports 1 x 300 W Power Supply

IBM xSeries 346 server single processor:

Rack-mount 2U 1 x Intel® Xeon™ Processor 3 GHz/800 MHz, 2 MB L2 Cache Processor 2 GB Memory 73.4 GB SCSI drive On-board 10/100/1000 Ethernet 3.5” 1.44 MB Diskette Drive 8X Internal DVD-ROM 1 x Serial Port 3 x USB Ports 2 x Ethernet Ports 1 x each of: SCSI, Keyboard, video and mouse ports 2 x 300 W Power Supply

IBM xSeries 346 server dual processor:



Rack-mount 2U 2 x Intel® Xeon™ Processor 3 GHz/800 MHz, 2 MB L2 Cache Processor 2 GB Memory 73.4 GB SCSI drive On-board 10/100/1000 Ethernet 3.5” 1.44 MB Diskette Drive 8 x Internal DVD-ROM 1 x Serial Port 3 x USB Ports 2 x Ethernet Ports 1 x each of: SCSI, Keyboard, video and mouse ports 2 x 300 W Power Supply

IBM xSeries 3650 server processor:

Rack-mount 2U 1 x Intel® Dual-Core Xeon™ Processor 2.33 GHz/1333 MHz, 4 MB L2 Cache Processor 2 GB Memory 2 x 146 GB SCSI drive (second drive is for RAID) On-board 10/100/1000 Ethernet 3.5” 1.44 MB Diskette Drive 8 x Internal DVD-ROM 1 x Serial Port 3 x USB Ports 2 x Ethernet Ports 1 x each of: SCSI, Keyboard, video and mouse ports 2 x 835 W Power Supply



Identify the operating system associated with each software element used within the proposed solution.

The 3Com VCX call control and application elements run on a hardened (highly secure) version of an industry-standard Linux kernel. This is a multi-threaded, multi-tasking, operating system that has proven suitable for enterprise networks. The 3Com-hardened Linux operating system is used for VCX, IP Messaging, and Convergence Application servers.

Describe the operating system hardening supported by the proposed solution.

The 3Com VCX IP Telephony solution operates on a Linux operating system with additional security measures on industry standard enterprise-grade servers. A choice of servers is available to provide the optimum reliability and performance characteristics based on your needs.

The VCX Linux operating system is used across all 3Com converged application servers in the solution, providing a uniform environment for operating Linux based software in a stripped-down and optimized manner. The VCX Linux operating system includes additional security measures such as disabling all ports, devices, and services that are not required by the VCX solution. 3Com’s VCX solution also employs internal automated defenses through an industry standard Linux firewall that gives granular access control over and protected access to VCX services, devices, and users.

In addition, remote access to a VCX server is limited to secure shell (ssh) and secure ftp (sftp) protocols. The VCX solution is architected such that security patch management is simple and recovery is always enabled from failed operating system upgrades.



Describe the equipment and interfaces that are used to connect to the Public Switched Telephone Network (PSTN).

The 3Com VCX IP Telephony solution uses digital media gateways and analog media gateways for connectivity to the PSTN and to PBX’s. These gateways provide cost-effective, scalable, and reliable connectivity to the PSTN and other switches that support a wide variety of TDM-based signaling interfaces, including T1/E1 with CAS/PRI/QSIG and analog lines.

The VCX IP Telephony system is fully standards based and supports common interface types for T1 (RJ-48), Ethernet (RJ-45), and POTS (RJ-11). All IP voice components within the boundary of the VCX IP Telephony solution seamlessly interface to the PSTN via a digital media gateway or an analog media gateway. The media gateway performs the necessary protocol and A/D media translations to enable this integration with the PSTN.

The gateways can be co-located with the VCX servers or at remote sites. This allows you to deploy the media gateways in locations that make sense economically throughout the enterprise; providing centralized PSTN access for savings on service provider costs, and/or distributed PSTN access for least cost routing. The 3Com VCX solution also supports interfaces to legacy PBX’s and switches using the digital media gateway.

3Com offers different VCX gateways that provide integration with the PSTN and PBX devices:

V7122 digital media gateway V6100 branch office modular gateway V7111 analog media gateway

These media gateways provide seamless SIP integration with the VCX solution, reliable signaling connectivity to PSTN and PBX devices, and support centralized administration and management.

VCX V6100 Branch Office Modular Gateway

The VCX V6100 branch office modular gateway provides the same digital T1/E1 capability as the V7122 digital media gateway, while being housed in a modular chassis that supports 1, 2, or 4 digital T1/E1 spans. This chassis can also be used to house separate CPU and disk modules that run the IP Telephony and IP Messaging software.



VCX V7122 Digital Media Gateway

The VCX digital media gateway provides economical connectivity between an IP network and legacy digital PBX systems using either channel associated signaling (CAS) or ISDN Primary Rate Interface (PRI) TDM signaling.

Designed in a compact 1U chassis, the VCX digital media gateway scales from 1 to 16 T1/E1 spans and can easily be managed by our VCX Enterprise Management System and a built-in web administration interface.

Some other features of the Digital Media Gateway include the following:

Voice Codec support Tone detection/generation Announcement detection/generation Echo cancellation Dynamic Jitter buffer support T.38 fax support Silence suppression Dual 10/100 Ethernet ports Redundant Power Supply Hot swappable boards SIP support

The V7122 Digital Media Gateway supports the following signaling and transport.

CAS PSTN Protocol Termination MF-R1: Wink Start, delay dial, immediate start, MFC/R2 numerous country variants;

unique script for each country variant ISDN PSTN Protocol Termination ISDN PRI: ETSI EURO ISDN, ANSI NI2, DMS Switch, Japan INS1500, QSIG Basic

Call, Australian Telecom, New Zealand Telecom, Hong Kong Variant, Korean MIC, and others IP Transport IETF RFC 1889, RFC 1890 RTP/RTCP Transport, TCP, UDP

The Digital media gateway supports SNMP v2, embedded web server, and VCX V7230 Enterprise Management System.



VCX V7111 Analog Media Gateway

The VCX analog media gateway links an IP network to analog devices and key systems in branch offices and other sites that would otherwise be isolated because of signaling incompatability.

Designed in an updated compact factor chassis, the analog FXO media gateway can be equipped with 4 or 8 analog lines, the analog FXS media gateway can be equipped with 4, 8, or 24 analog phone/fax/modem ports, and both FXO and FXS gateways can easily be managed by our VCX Enterprise Management System. All but the 24-port version utilize RJ-11 connections to support analog fax machines. The 24-port model has a standard 50-pin Telco connector and is typically cross connected to a punch-down block.

In addition, an analog “combination” gateway is available in either a 2 FXO/2FXS model or a 4 FXO/4 FXS model. These analog combo gateways provide a cost effective solution for small remote locations by requiring only 1 unit for both functions.

Some other features of the Analog Media Gateway include the following:

FSX/FXO (4 and 8 channel) support 2-24 analog RJ-11 ports Internal Power Supply Message Waiting on FXS phones Dual 10/100 Ethernet ports Autodetect and Switchover call processor capabilities SIP support

The V7111 Analog Media Gateway supports the following signaling and transport.

2-Channel and 24-Channel: FXS loop-start 4-Channel and 8-Channel: FXS and FXO loop-start In-Band: DTMF (TIA 464B), MFR1, MFR2, AC15, SS4, SS5; user-defined and call

progress tones

The Analog Media Gateway supports VCX V7230 Enterprise Management System, BootP, DHCP, TFTP and Syslog.



How are analog phones and devices such as fax machines connected to the system?

3Com provides standalone analog FXS media gateways in 2-, 4-, 8-, and 24-port versions. All but the 24-port version utilize RJ-11 connections to support analog phones. The 24-port model has a standard 50-pin Telco connector and is typically cross connected to a punch-down block.

When a call is destined for an analog phone, 3Com’s VCX Call Processor will forward a SIP INVITE message to the appropriate FXS media gateway. Assuming the analog port is in a ready state, the media gateway responds with a 200 OK message and the voice path is established between the calling party (IP phone or ingress media gateway) and the analog port.

When an analog phone originates a call, the FXS media gateway issues a SIP INVITE message to the VCX Call Processor in the same way an IP phone would.

Does your system support Power Failure Transfer Station (PFTS)?

3Com supports lifeline services on each analog gateway by automatically cutting through a single analog line in case of power failure.

Port 4 can act as a lifeline port on 4 or 8 port FXS gateways and the combo FXO/FXS gateways. 24 Port FXS gateway do not support the lifeline feature.

Does the solution utilize proprietary signaling to IP handsets or to equipment that interfaces with the PSTN?

The 3Com IP Telephony solution uses SIP signaling to communicate with IP handsets, analog gateways that communicate with analog phones, and to analog and digital gateways that communicate to the PSTN or PBX.



Describe how the proposed solution supports tele-workers with soft phones and IP phones.

The IP Tele Commuter module is a component of the solution that allows remote employees to use an IP phone or convergence center client over the internet via a cable or broadband connection, accessing the same features as users at HQ. The IP Tele Commuter module allows for secure access and solves the NAT traversal issue that is inherently caused by remote employees being on a private network and attempting to access the corporate network using the internet.

In addition to the various media gateways the VCX also includes as an option a Telecommuting Module for managing remote phones connected to the system. The Telecommuting Module is a device which processes traffic under the SIP protocol (see RFC 3261). The Telecommuting Module receives SIP requests, processes them according to the configured rules, and forwards them to the receiver.

The Telecommuting Module connects to an existing enterprise firewall through a DMZ port, enabling the transmission of SIP-based communications without affecting firewall security. SIP messages are then routed through the firewall to the private IP addresses of authorized users on the internal network. The Telecommuting Module can also be used as an extra gateway to the internal network without connecting to the firewall, transmitting only SIP-based communications.

The 3Com IP Telecommuting Module extends the benefits of the 3Com Converged Application Suite to users connecting to the enterprise network from remote locations. With the module, home office workers, traveling employees, and other authorized users can securely access the IP telephony system and take advantage of a wealth of communications functions based on the Session Initiation Protocol (SIP).


CorporateNetwork

Standalone

Traveling User

HomeUserNAT

HotelRouter

HotSpotUser

Internet

Presence

Messaging

Contact Center

Conferencing

DMZ

InternetInternet InternetInternet

DMZ + Direct

InternetInternet

VCX

IPTelecommuting

Module

TelecommutingModule

TelecommutingModule

TelecommutingModule

TelecommutingModule

TelecommutingModule

TelecommutingModule


Can our remote locations survive and continue with full feature functionality if the central call processing system is unavailable for any reason (such as a network failure or server problem)?

The 3Com VCX architecture is designed to operate in a centralized or distributed fashion, scaling from 200 phones to thousands of phones in a logical, cost-effective manner. The VCX solution provides redundancy at many levels, ensuring high availability even when individual components fail or WAN connectivity is lost. To meet the needs of distributed enterprises, 3Com created the voice boundary routing architecture to provide the scalability, flexibility, and reliability that is needed to support all the users across the enterprise.

As illustrated in the figure below, the 3Com VCX voice boundary routing architecture deploys call processing in a hierarchical manner. In this architecture, VCX call processors and media gateways are deployed at branch offices, which are connected to a regional office call processor for redundancy, centralized routing, messaging, applications, and global directory services.

A regional office provides call processing for HQ users, supports many branches, can provide centralized PSTN connectivity, and supports either local or geographically dispersed redundancy. In addition, the regional office often provides centralized applications, such as messaging, conferencing, instant messaging, and presence.

The flexibility of the 3Com voice boundary routing architecture allows branch offices to deploy cost-effective media gateways that provide analog or digital connectivity to the PSTN for local calls, emergency calls, and survivability. The call processors deployed at branch offices are designed to support a specific set of users, with their own dial plans and routing configurations, yet still have redundancy back to the central site, a local and global user directory, and inter-regional and inter-branch dialing.


Regional Office 1

Branch Office1-1

WANWAN

Branch Office1-n

Branch Officen-1

Branch Officen-1

Regional Office nPSTNPSTN

PSTNPSTN

Centralized Administration and Managementwith Global Directory Services

Centralized Administration and Managementwith Global Directory Services


The branch office call processors are made redundant with the primary server of the regional office. If the messaging application is present at the branch offices, automated archival of messaging data to the regional office messaging system is performed. In addition, the VCX supports remote sites connected to regional or branch offices (for call processing) where IP phones and gateways are present, but call processors are not.

The 3Com VCX Voice Boundary Routing architecture supports the ability to add multiple regional offices to the system, all connected together with a global directory and centralized administration. This flexible architecture allows you to design and build a reliable, scalable converged communications system that provides true multi-site, enterprise-wide features using centralized administration.

Describe how features function transparently for users communicating within a distributed system.

The VCX solution supports an enterprise-wide, multi-site architecture that offers superior feature survivability at branch offices during WAN outages. Multiple regional offices can be connected together to provide superior scalability while providing a centralized view into administration, centralized management, global user directory, and inter-region call routing.

Based on the VCX’s powerful multi-site voice boundary routing architecture, the calling features of the VCX are designed to be multi-site in nature. Individual users are provisioned off their home regional office.

When configurations are saved at the central call processor, are they automatically propagated to survivable remote locations?

The 3Com VCX solution supports true multi-site deployments, where multiple regions can contain multiple branches. Call processing is local to each regional and branch office, yet all function together as a cohesive system. Note that a regional office is considered one site, even if its servers are geographically separated at two different locations.

Describe how survivable remote locations are protected from both WAN failures and router failures.

In the event of a WAN failure, a branch office VCX call processor provides full feature transparency and is equipped to enable local and long distance PSTN calls. In the event of a failure of the local VCX call processor, local IP phones and gateways automatically register with its secondary call processor located at the regional office.

In the event of a failure of a regional VCX call processor, the local VCX call processor continues to operate normally. In the event when neither the primary or secondary VCX call processor is available, the VCX solution supports PSTN survivability for inbound calls to an IP phone and outbound calls to PSTN from any IP phone configured via DHCP.

Describe the survivability characteristics of remote locations that do not have a call processor or when communications to all call processors is lost.

The VCX provides remote survivability in case of server failures, LAN, and WAN failures.



If a phone or media gateway does not find its primary server, it looks for its secondary server after its registration has expired or it tries to initiate a new call. If that fails, the phones and media gateways can be configured to provide support for both inbound and outbound calling (outbound only with DHCP-enabled devices).

Remote survivability works the same way in both analog and digital media gateways:

To handle inbound calls in the case of a WAN failure, the “Tel to IP” Routing table in the gateways needs to be populated with the phone number and the corresponding IP address – 50 such entries are possible in a gateway.

PSTN outbound calling can be achieved by directly dialing out through the gateway.

Extension to extension dialing will be possible only by dialing the DID and the call will be routed via the gateway.

Explain how inbound/outbound local calls and emergency calls are handled at the remote locations.

For a solution that involves multiple locations, it is often desirable to have centralized redundant T1/PRI’s that handle the majority of incoming calls (for example, from a toll-free number), handle all outgoing long distance calls, and can provide emergency services. At remote locations, analog media gateways provide connectivity to the CO using analog POTS lines for local incoming calls, local outgoing calls, and emergency services.

Incoming calls to media gateways at remote locations can be handled in several ways, including routing directly to any phone in the enterprise (local or remote) based on DID or phone number, to a hunt group, to a voice mailbox, or to an auto-attendant. Multiple Call coverage points are also available for each phone and hunt group based on a time of day, day of week, event, and holiday basis. Auto-attendants can be designed to provide a local dial-by-name directory if required.

The System i IPT solution supports the ability for all users across the enterprise to have their long distance calls be placed through a specific set of media gateways (for example centralized at HQ and backup locations). The media gateways at the remote locations are often used simply for outgoing local calls originating from the location and for emergency services. The System i IP Telephony solution also supports the ability to provide Least Cost Routing (LCR) which allows a phone call to be routed across the IP network to a media gateway that is within the local calling area of the destination number, reducing long-distance costs.



2.3 Redundancy and Reliability

The system must be designed with no single point of failure for the entire system including servers, software, and interfaces to external networks. Describe how the proposed solution provides system-level redundancy in a geographically separated manner.

The VCX is architected in a fully redundant fashion to preclude “total system failure.” Call control and messaging servers will failover gracefully to backup servers. When the primary server is restored to service, which can be done automatically based on the nature of the failure, control will pass gracefully back to the primary server. In the case of IP Messaging, after automatic recovery of a failed server, the Intelligent Mirroring feature automatically re-synchronizes the two IP Messaging servers.

Customer specific data is protected through server failures. The time required to restore a properly configured server to operation is will range between 2-5 minutes based on user population.

In the event of a WAN failure, a branch office VCX call processor provides full feature transparency and is equipped to enable local and long distance PSTN calls. In the event of a failure of the local VCX call processor, local IP phones and gateways automatically register with its secondary call processor located at the regional office.

In the event of a failure of a regional VCX call processor, the local VCX call processor continues to operate normally. In the event when neither the primary or secondary VCX call processor is available, the VCX solution supports PSTN survivability for inbound calls to an IP phone and outbound calls to PSTN from any IP phone configured via DHCP.

The VCX supports call control redundancy including database replication of two or more nodes. On the messaging side, the VCX supports redundancy in a feature called Intelligent Mirroring, which provides synchronization of messaging configuration and mailbox data to a hot standby messaging server.

The 3Com VCX ensures high availability using a primary/secondary redundancy architecture that replicates data in real time, uses little bandwidth to accomplish this, and provides transparent failover for users, VCX applications, media gateways, and phones. The Call Server and Authentication & Routing Server are replicated independently of each other, maximizing resiliency of the VCX software architecture.



Redundancy of selected survivable branch office locations is required. Describe how the proposed solution supports redundancy of call processing servers located at survivable branch office locations.

The 3Com VCX solution provides the ability to deploy a distributed network of branch offices with call processing that are redundant to a regional office. Branch offices are configured with their own routing, authentication, and user information in addition with the necessary authentication and routing information to communicate with the regional office. With a low-bandwidth replication solution, the redundancy does not adversely impact WAN utilization.

A VCX multi site system consists of one or more regional offices, with each regional office optionally consisting of one or more branch offices. The regional office is itself redundant, either locally within one rack, or geographically separated in another room, building, or location over the WAN. The call processors at the branch offices are made redundant to the regional office by using the same replication scheme used by the regional (or single site) redundancy scheme. All branch offices use the regional office primary VCX call/data server as their secondary server. In this manner, all branch office databases are replicated at the regional office primary VCX server.

Regional offices are also configured with their own routing, authentication, and user information along with the necessary authentication and routing information to communicate with all other regional and branch offices.

The 3Com VCX solution allows you to administer and provision all sites from a web browser based centralized management console. Corporate administrators can access all sites from the regional office and branch administrators can be given access to their own local branch.


IP Telephony

Primary Site Secondary Site

LANLAN

Replication

LANLAN

WANWAN

Branch Offices

LANLAN

IP Messaging

IP Telephony

IP MessagingIntelligent Mirroring

Satellite Offices

LANLAN

Replication


In either case, moves/adds/changes are replicated across the WAN to the assigned redundant server.

Failovers to redundant servers or gateways must be automatic and must not require any action by users. Established calls must not be dropped during failover. When unavailable equipment becomes available, the system must automatically be restored to its redundant configuration. Describe how your solution provides automatic failover and automatic recovery of redundant equipment.

To ensure high availability, the 3Com IP Telephony solution supports an inherent redundant capability. One of the call processors is identified as Primary and the other is identified as Secondary. There is no voice path established to the IP Telephony call processors, only SIP signaling, requiring minimal IP bandwidth for call setup. The voice path is established between end points and does require sufficient IP bandwidth.

In case of Primary call processor connectivity failure, users will failover to the Secondary call processor. The failover to the Secondary call processor occurs automatically the next time an inbound call arrives, an action occurs at the gateway or phone (i.e. placing a call), or when a keep-alive response is not received from the currently registered call processor. Active calls stay live and do not get dropped. There is no loss of functionality after switchover to the Secondary call processor.

3Com phones work in survivability mode when there is multiple point of failure. Phones go to gateways directly if the call processors fail, going into a limited support mode where inbound calls can be received and outgoing calls can be placed (outgoing calls only when phone is configured via DHCP).

The System i IP Telephony solution also provides a robust set of routing features, including alternative routes. The IP Telephony call processor use alternate routes to other gateways if a route to a particular gateway is not found. The IP Telephony call processor continues to monitor the gateway, automatically allowing traffic to flow through the gateway upon its availability.


PRIMARY

Real Time Replication

SECONDARY


SECONDARYPRIMARY

SECONDARYPRIMARY


SECONDARYPRIMARY



2.4 Scalability

Describe how your solution provides scalability in call processing capacity, PSTN gateway capacity, messaging mailbox capacity, and user capacity as the number of physical access ports and number of users grows over time.

The 3Com VCX architecture provides significant flexibility in the configurations that can be used at regional and branch offices. The ability to distribute the primary software components across multiple servers and to have them replicate each other is the key factor that makes the VCX solution a superior choice. This same ability allows the VCX to scale from one configuration to the next simply by adding servers, server licenses, and performing a managed migration of the software and databases to the new configuration.

Satellite Offices

Satellite offices are typically remote locations that do not have call processing servers, but do have PSTN survivability. Scalability at remote locations includes quantity of PSTN lines, analog devices, and IP phones. Bandwidth over the WAN will be required as number of users increases at satellite offices (this is the case for all types of remote locations). Satellite offices will have analog or digital media gateways that provide local PSTN access. Users at the satellite offices will be able to talk to users at other sites or VM over IP network. The analog and digital gateways also provide PSTN survivability and emergency services for the satellite office they are installed in.

Branch Offices

At a branch office, the VCX solution typically consists of a V6x00 branch office controller and IP phones. The V6000 model provides 4 FXO and 2 FXS ports in addition to a server blade that hosts the same software that runs at the regional office, only scaled down to branch office size. The V6100 model is a modular unit with optional CPU/disk modules and T1/E11/2/4 span modules. The V6x00 models support an optional second hard drive (for RAID) and an optional redundant power supply.

For branch offices, the V6x00 server typically is either an “All-In-One” or a “SoftSwitch” server, depending on whether messaging is local or global. This configuration is made redundant by using a VCX regional office primary server as its secondary server.



In addition, depending on scalability requirements, a V7005 or V7205 server and media gateways can be used in place of the V6x00.

Small Regional Office

A VCX configuration for a small single site solution typically consists of 2 servers, multiple media gateways, and phones. This is a redundant configuration that does not support any branch offices.

This configuration also provides a cost effective solution for a small single site with multiple remote sites, where the redundancy and SIP-based architecture of the VCX is desired. Both servers are installed with the “All-In-One” software configuration. In addition, the primary server has the Call Record Server software installed.

Medium Regional Office

A VCX configuration for a medium single site typically consists of 4 servers, multiple media gateways, and phones. This is a redundant configuration which supports branch offices.

Two of the servers run the IP Messaging application, and two of the servers run the IP Telephony application. In addition, the IP Telephony primary server has the Call Record Server software installed. This configuration can be scaled to a large multi site with branch offices by distributing the VCX software on additional servers.

Large Regional Office

A VCX configuration for a large regional office or large single site typically consists of 6 or more servers (depending on exact scalability requirements), multiple media gateways, and phones. This is a redundant configuration which supports branch offices.


All-In-OneCDR Server

All-In-One-> IP Telephony + IP MessagingRegional Office

CallPCallP

Auth & Dir Auth & Dir

MMU MMU

MSU MSUCDR Server

I P Telephony

IP Messaging

Regional Office

IP MessagingIP Messaging

IP TelephonyCDR Server IP Telephony

Regional Office


For enterprises with large numbers of users, the 3Com VCX IP Telephony solution allows for multiple regional offices to connect together to provide this level of scalability without sacrificing feature transparency or reliability. Each regional office supports a sub-set of the total user base, while allowing for a global directory, multi-site calling, user mobility, centralized administration (each regional office is provisioned separately but accessible centrally), all in a secure reliable manner.

2.5 Interoperability

Describe your philosophy on open architecture and your ability to support other vendors’ equipment.

Since 3Com's founding in 1979 and creation of the Ethernet standard more than 30 years ago, the world has embraced 3Com’s vision of pervasive networking:

Every personal computer contains a network connection; Businesses are fundamentally built around a flow of information carried by a network,

and; Enterprises large and small are increasingly adopting "converged" networks that

include Internet Protocol (IP) telephony technologies to achieve significant cost savings and dramatic new functionality and features that enhance the bottom-line.

3Com has led the industry in defining many of today’s IP-based standards with its strong background in IP product development. Today, the open architecture of the VCX allows it to be a cost-effective solution that inter-operates well with other IP LAN/WAN infrastructure and SIP devices. 3Com is committed to converged voice and data applications using open software architecture and standard protocols. Our philosophy is to work with integrators and third-party vendors to provide voice-specific applications. These applications include: voicemail, unified communications, conferencing, call center, presence, instant messaging, and others.

3Com believes that the most open, interoperable architecture will be the most beneficial, cost-effective architecture to the customer. By adhering to open standards, a 3Com VCX solution ensures the customer is not tied to a sole vendor situation for all their IP needs.

The ability to choose different products that best meet our needs is important to us. Describe how 3rd party SIP-based solutions are tested for interoperability with the proposed solution.

The 3Com solution leverages open standards, allowing customers to exercise choice with IP handsets, media gateways, and applications. Many 3rd-party components and applications have been certified to interoperate with 3Com’s VCX solution, giving customers the option of deploying best-in-class solutions without being locked into a particular vendor technology.

Since VCX is a pure SIP solution, your enterprise is not locked into a single vendor solution. You can select the handsets, gateways, and backend services that best meet your needs. The 3Com solution leverages open standards, allowing customers to exercise choice with IP handsets, media gateways, and applications. Many 3rd-party components and applications have been certified to interoperate with 3Com’s VCX solution, giving customers the option of deploying best-in-class solutions without being locked into a particular vendor technology.



Does the proposed solution support 802.11a/b/g? Describe the WiFi handsets supported by the proposed solution.

The 3Com 3108 WiFi phone is planned for availability in an upcoming release. This device features the first SIP based color flip phone weighing 89 grams. It is IEEE 802.11g/b compliant and supports WEP. It is designed to achieve a long battery life of 4 hours talk time and around 100 hours standby time. The phone has an embedded web browser and multiple ESSID support.

Describe your IP signaling capabilities and their conformance to standards. Clearly identify open or international standards versus proprietary standards. (Note: standards supported by a single vendor do not qualify as open or international, regardless of market share. They are, by definition, proprietary.)

The 3Com VCX IP Telephony Module supports the telecommunication and networking standards listed in the following table.

Standard AvailabilityG.711 Fully SupportedG.726 Supported by gatewaysG.728 Supported by gatewaysG.729 Supported (except for TTS)G.729a Supported (except for TTS)H.323 V2 Supported via 3rd party gatewayQ.931 Fully Supported by Digital Media Gateway802.1d Fully Supported802.1p (LAN Prioritization) Fully Supported802.1q (VLAN) Fully Supported802.3af (PoE) Fully SupportedSIP RFC 3261 Fully SupportedSNMP v2 Fully SupportedFAX - Group 3 Fully Supported by gatewaysFAX - Group 4 Fully Supported by gatewaysT.38 Fully SupportedIP Precedence Fully SupportedDiffServ Fully SupportedWeighted Fair Queuing Fully SupportedRED Fully SupportedWeighted RED Fully SupportedRTP Fully SupportedRTCP Fully SupportedPolicy Based Routing Fully SupportedIPv6 OS supports; full support on RoadmapH.263 Fully SupportedTCP/IP Fully SupportedUDP/IP Fully SupportedDHCP Fully Supported



DNS Fully Supported

Does your system employ proprietary protocols for the telephones to learn their voice VLAN or is an industry standard used?

The 3Com IP phones used with the VCX solution use industry-standard protocols such as DHCP to discover their VLAN and other IP and VCX configuration data.

Does the proposed solution utilize a proprietary method to power the IP Phones, or are industry standards supported? Describe the support for Power over Ethernet, including the 802.3af specification.

All 3Com PoE switches are based on industry standard 802.3af. 3Com phones include 10/100 Base-T switch ports and standards-based 802.3af Power over Ethernet.

Describe your support of out-of-band dual tone multifrequency (DTMF) signaling over IP.

3Com’s VCX supports RFC 2833 DTMF.



2.6 Data Network

Identify the data networking requirements to support quality of service for the proposed solution.

The 3Com solution uses 802.1p User Priority/Traffic Class to achieve QoS.

802.1P tagging is done by all the sets Servers and Gateways can be tagged by configuring the switch to which they are

connected

QoS is supported by 802.1q tagging via either a DHCP configuration or manual configuration on the phone. This enables voice traffic to go on a separate VLAN from other traffic which can be prioritized by the routers/switches.

802.1Q tagging done via DHCP or on the phones VCX Servers and Gateways can be tagged by configuring the switch to which they

are connected

3Com recommends a core-to-edge network utilizing separate VLANs to segregate voice traffic from data traffic.


Core Switches

VCX IP TelephonyServers

IP Conferencing/PresenceServers

IP MessagingServers

Edge PoE Switches

Edge Router

WAN

Core VLAN for Servers

Two Trunked VLANS betweenEdge switches and Core switches/routers:

• Default Untagged VLAN (for PC’s)• Tagged Voice VLAN (for IP phones)

Edge PoE Switches

Analog MediaGateways

Digital MediaGateways

Voice VLAN for RTP Devices

At The Edge:

At The Core:

• Separate interfaces for traffic• Core VLAN for servers• Voice VLAN for RTP Devices

• Switch Ports are configured for default VLAN and Tagged Voice VLAN

Voice VLAN

Core Routers


Does the solution require a specific vendor’s data networking equipment?

The 3Com IP Telephony solution functions on any vendor’s data networking infrastructure that supports standards-based quality of service and power over Ethernet.

The 3Com solution uses 802.1p User Priority/Traffic Class to achieve QoS.

802.1P tagging is done by all the sets Servers and Gateways can be tagged by configuring the switch to which they are

connected

QoS is supported by 802.1q tagging via either a DHCP configuration or manual configuration on the phone. This enables voice traffic to go on a separate VLAN from other traffic which can be prioritized by the routers/switches.

802.1Q tagging done via DHCP or on the phones VCX Servers and Gateways can be tagged by configuring the switch to which they

are connected

Describe the ways we can monitor Quality of Service of VoIP calls.

The use cases that the QoS monitoring feature are designed around are:

Before and after measurements of layer-2 configuration changes to weigh effect of changes

Identifying faulty endpoints or routes Identifying periods or network episodes with sub-standard voice quality

SLA enforcement, network capacity planning info. are NOT features we claim to support

View QoS statistics feature as a troubleshooting/data gathering tool

The QoS monitoring feature implemented on the VCX system covers 3Com IP phones as endpoints (310x), with the administrator being able to use EMS to configure:

One or more monitored endpoints/subnets Thresholds for QoS stats generation by the phones on those monitored subnets (not

every call will generate a QoS record) Alarm thresholds for notifications to be issued based on QOS parameters View generated statistics using SNMP Soft-phones and convergence clients are NOT supported QoS monitoring is implemented across three components:

o VCX Call Processoro 3Com IP Phoneso XML Accounting Server

A call processor can only process statistics from phones that are registered to it. If monitored endpoints are setup on the primary and secondary Call Processor, only the Call Processor that the phone is currently registered to will recieve the QoS reports from the phones



o The configuration information (monitored endpoints, thresholds etc.) is persistent across system reboots and upgrades

o The Endpoint Statistics and Call History tables available for viewing from EMS do not persist across reboots of the system, upgrades, or restarts of the XML Accounting Server

o No QoS data is lost – it is all available in the XML QDR (Quality Detail Records) stored by the Accounting Server



2.7 IP Phones

Are the IP phones in the proposed solution RoHS compliant?

The new models 3101B, 3102B, 3103B, and 3105B are RoHS compliant versions of the 3Com IP phones.

Describe your basic phone set model, including the number of system appearances, buttons, display, and features of the phone.

3Com’s 3101 Basic (Without Speaker) IP Phone is ideal for day-to-day office use when a microphone on the phone for hands-free use is not required. The 3Com 3101 Basic (Without Speaker) IP phone supports the following attributes:

line appearances programmable buttons with lights fixed feature keys

o Volume up, volume down, mute, hold, voice mail 2-line pixel display soft keys for use with display menus 4-way display control Large message waiting lamp No speaker button Speaker for paging only Headset jack Articulating stand/wall mount Link Security 802.1af compliant Power over Ethernet 802.3af compliant Dual switched 10/100 Mbps uplink ports Wideband Audio HW-Ready (Handset Only)



Describe your business phone set model, including the number of system appearances, buttons, display, and features of the phone.

3Com’s 3102 Business IP phone is ideal for power users requiring speakerphone and one button access to multiple features or line appearances. The 3102 Business IP phone includes the following attributes:

5 line appearances (3 own, 2 bridged) 18 programmable feature/line buttons with lights 5 fixed feature keys with replaceable faceplate for localization

o Speaker buttono Redialo Conferenceo Transfero Hold

6 fixed feature keyso Volume upo Volume downo Muteo Hands Freeo Forward to Voice Mailo Voice Mail

soft keys for use with display menus 2-line pixel display 4-way display control Large message waiting lamp Acoustic chamber for full-duplex speakerphone Unidirectional microphone Definable language faceplates Headset jack Articulating stand/wall mount Link Security 802.1af compliant Power over Ethernet 802.3af compliant Dual switched 10/100 Mbps uplink ports Built-in support for wideband audio



Describe your manager phone set model, including the number of system appearances, buttons, display, and features of the phone.

3Com’s 3103 Manager IP phone is ideal for managers or executives. The 3103 Manager IP phone will be available at VCX 7.0 GA. The 3103 Manager IP phone includes the following attributes:

5 line appearances (3 own, 2 bridged) 8 programmable feature/line buttons with lights 5 fixed feature keys with replaceable faceplate for localization

o Speaker button o Redialo Conferenceo Transfero Hold

6 fixed feature keyso Volume upo Volume downo Muteo Hands Freeo Forward to Voice Mailo Voice Mail

10 soft keys for use with display menus large screen display 4-way display control Large message waiting lamp Acoustic chamber for full-duplex speakerphone Unidirectional microphone Definable language faceplates Headset jack Articulating stand/wall mount Link Security 802.1af compliant Power over Ethernet 802.3af compliant Dual switched 10/100/1000 Mbps uplink ports Built-in support for wideband audio



Describe the attendant console available with the system.

The 3Com 3105 Attendant Console gives workgroup administrators and receptionists a flexible and intuitive tool for handling calls and viewing phone status for up to 100 users. To service larger locations with hundreds of users, multiple consoles can be connected in parallel.

In addition to 50 programmable buttons with functionality that can be doubled with a press of the SHIFT key, the console offers four additional buttons reserved for frequently used features. The 3105 model also supports Direct Station Selection and Busy Lamp Field (DSS/BLF) functions, CO line appearances, call park zones, and the 802.3af PoE standard.

The 3Com 3105 Attendant Console gives workgroup administrators and receptionists a flexible and intuitive tool for handling calls and viewing phone status for up to 100 users.

The 3Com 3105 attendant console allows receptions or departmental assistants to perform standard console functions such as receiving calls, viewing the status of incoming calls, and transferring calls.

• 100 Speed dials– Busy line indication– Emergency Call indication

• Configurable button mappings– Transfer– Attendant Serial Transfer– Hold– Park– Conference



Describe the SIP-based soft phones available with your solution.

The 3Com Convergence Center Client is a Java-based application that allows real-time communication using the Session Initiation Protocol (SIP). With the 3Com Convergence Center Client, you can make voice calls from your desktop PC to other computers or to telephones on the PSTN. You can add video to the conversation, plus exchange instant messages, share desktops, and exchange web pages. You can also hold multi-party conferences, sharing voice, video and other applications in a collaborative environment.

The 3Com Convergence Center Client can be installed on computers with Windows 2000 and XP operating systems.

Here are some of the things you can do using the 3Com Convergence Center Client:

Create a Buddy List of your friends and coworkers who use the VCX. See when your buddies are online and available, then call them or send them instant

messages. Have a video/voice conversation using your computer microphone, speakers, and camera. Click to migrate from one media (such as voice) to another (such as video). Start a desktop sharing session and optionally take command of another person’s desktop. Share web pages with another user. Participate in multiparty conferences which include voice, video, instant messaging, and

desktop sharing services. Drag and drop your buddies into a conference. Display multiple participants during a video conference. Display your own, local video window. Receive Caller ID messages. Transfer calls; place calls on hold. Enter DTMF dial tones using the integrated dial pad. Quickly and easily connect to the Voice Mail system to listen to your messages. Receive a message waiting indication when you receive a new Voice Mail message.


>Presence

>Instant Messaging

>Voice

>Video

>Desktop Collaboration

>Voicemail View and Access

>Call history— Outgoing calls

— Incoming calls

— Missed calls

>Supports re-dialing from log

>Sort by name, number, time, duration

>Call history— Outgoing calls

— Incoming calls

— Missed calls

>Supports re-dialing from log

>Sort by name, number, time, duration

>Contact List>Import from CSV

>Create contacts

>Multiple nos.>Buddy or not>Dial direct from contact list

>Contact List>Import from CSV

>Create contacts

>Multiple nos.>Buddy or not>Dial direct from contact list

Availability Controls

Presence & Call

Active sessions

Media

Voicemail

Availability Controls

Presence & Call

Active sessions

Media

Voicemail


What are the PC requirements for the SIP softphones?

These are the minimum hardware and software requirements for the 3Com Convergence Center Client:

Microsoft Windows requirements

For Microsoft Windows, the minimum requirements are:

Microsoft Windows 2000, or XP Home or Professional 256 MB RAM Without video: Intel Pentium III 750 Mhz or higher; With video: or Intel Pentium III 1 Ghz or higher.

Hardware compatibility

The 3Com Convergence Center Client is compatible with the following hardware components:

Full duplex sound card and associated software driver Microphone and headset USB web camera supporting CIF/QCIF Recommended web cameras: Logitech QuickCam Pro 3000 LogiTech QuickCam Pro 4000 LogiTech ClickSmart 510

Describe how a large number of soft phone clients can be auto-configured.

The 3Com Convergence Center Client supports a bulk auto-configuration template that is downloadable by Convergence Center Clients from a server. The template provides all of the client configuration information except the user’s SIP address.

After the Convergence Center Client is installed and started for the first time, it prompts for the URL address of the configuration file, which typically has this format:

http://<server name or IP address>/MasterConfig.xml



Describe the SIP-based software attendant console supported by the proposed solution.

The software attendant console is an optional third party package that is fully interoperable with the System i IP Telephony solution. It provides these key features:

Real-time VCX device state (idle, busy, ringing, connected) Modify any VCX device (DND, Call Forwarding, Calling party information, Caller ID,

and Caller Name) Drag & Drop call processing Works with ANY VCX multi-line set Call processing features

o Answer Call, Hang-upo Transfer Call (supervised / blind)o Hold / UnHold, Send to Voice Mail

Can a user control their IP phone from the soft phone client? If so, what features are supported?

Enables Convergence Client to control phone

Initiate call on convergence client Use hard phone for call Dial from contact list and call logs Using video if required

Identify the languages that are supported for phone displays. Can a specific language be configured on a per-group or per-user basis?

English (US) English (UK) French (France) French (Canadian) Chinese (Mandarin) Spanish (LAT)



Describe the options for powering IP phone sets.

3Com phones include 10/100 Base-T switch ports and standards-based 802.af Power over Ethernet. Optionally, 3Com IP phones can use local power. If Power over Ethernet is not available, phone power supply “bricks” must be purchased separately for each 310x phone.

Describe the options for connecting the IP phone sets to the LAN.

The VLAN may either be set manually or by using DHCP option 184 settings.

If both a phone and PC are connected to the same port on the switch, all traffic is untagged and the voice and data traffic are mixed together. Auto VLAN cannot be used in this situation either, as the PVID of the port is decided by the OUI of the 3Com phone.

If a 3Com phone supports tagged voice traffic, this feature can be used because the modification of PVID has no effect on tagged voice traffic. When the PC and Phone have been configured on different VLAN, the traffic is isolated and cannot be seen by each other.

Can IP phone sets share existing Ethernet ports with data devices or do they require separate Ethernet ports?

The 3Com phones contain a switch chipset that supports 10/100 Mbps. The 3101 and 3102 models all have a dual switched 10/100 Mbps uplink ports; one for the phone, the other for a PC. The 3103 model has a dual switched 10/100/1000 Mbps uplink port.

The data port on the 3101, 3102, and 3103 phones is a 2-port active switch that requires power. If power is lost, any device plugged into the phone’s second port will lose connectivity.

Do any of your IP phone sets support gigabit to the desktop?

3103 models all have a dual switched 10/100/1000 Mbps uplink ports

Do IP phones require manual intervention to upgrade phone software?

With the VCX IP Telephony solution, 3Com’s SIP-based IP phones support automatic detection of phone software update version. This allows software upgrades to be delivered automatically to phones without the need to reboot phones. Upgrade detection is polled-based on 30-minute cycles.

Describe the user mobility options. Can users log on/off a phone so that two (or more) users can use the same phone, but with different line options and features?

The system administrator determines whether the end user can move their station. Phones/soft clients can be locked to a physical switch port (using MAC address) or to a specific IP subnet. Otherwise, end users can move their phones to any location that has IP access to the user’s primary or secondary VCX call processor, regardless of geographic location.





2.8 System Features

The proposed solution must provide the ability to route long distance calls over the appropriate, usually least costly, trunk group via public or private networks, IP or traditional.

To assign cost to routes and find routes based on lowest route.

The directory service, in conjunction with the authentication service, supports sequential route determination on a user-by-user (or wildcard) basis. This allows different call treatments for different users including least-cost routing, time-of-day routing, ingress/egress routing, routing to other destinations (call coverage), routing to voicemail, etc.

The VCX supports least cost routing through the priority setting of each route. The system will always try the highest priority route first. The VCX is able to screen up to 28-digits in order to determine how the call is to be routed.

Any number of routes may be configured.

The proposed solution must support the ability to route calls based on ANI/DNIS/CLID incoming call information.

The 3Com VCX IP Telephony solution supports Direct Inward Dialing (DID) on the digital media gateways. DID connects calls from the PSTN directly to a dialed extension number without attendant assistance. Specialized DID trunk circuits from the service provider are required to implement this feature.

The proposed solution must support the ability to route calls to alternate route points in cases of congestion or failure of a device interfacing with the PSTN.

The VCX supports the ability to provide alternate routes to reach the same endpoint. 3Com’s VCX solution supports load sharing and alternate route selection when multiple routes can be chosen for a call.

The load information can be updated from the endpoints through SIP REGISTER messages, so that the system always chooses the least loaded termination gateway first given same priority. The solution can also try the alternate route once it detects the primary one is not available, busy, or in error condition.

Describe the ability of the proposed solution to support an unlimited number of music on hold sources, and how these sources can be assigned on a per-user basis.

The 3Com VCX solution provides a scalable, efficient, and flexible Music on Hold feature. Music on Hold (MOH) allows callers to hear a particular recording continuously while on hold. The VCX Music On Hold feature allows administrators to assign specific MOH files to different groups of users on a per-phone basis. The VCX solution supports an unlimited number of MOH sources.



The MOH sources are stored at the 3Com IP Messaging module. When a caller is to be placed on hold, the VCX redirects the call to the IP Messaging module at a particular extension. The extension given to the IP Messaging module identifies the MOH source to play.

The MOH sources are downloaded to IP Messaging in .wav format using sftp or the 3Com Enterprise Management Suite. Once the wav file is downloaded to an IP Messaging server, it is converted to both G.711 and G.729 formats required for play by IP Messaging. The IP Messaging Operations and Administration Guide and the EMS VCX User Guide details the procedures for creating and downloading MOH source files.The directory on the IP Messaging server where the MOH files are located is /usr/app/app.dir/speak.vox, and the MOH files must be named with lower case and fewer than 8 characters.

Music On Hold is implemented using IP unicast and does not require additional bandwidth. In addition, the Music On Hold feature is implemented in the VCX call processor instead of the phone, providing improved functionality and feature interactions:

There are no CDR’s generated for Music On Hold call legs. Music On Hold functionality is supported for all endpoints, including FXS ports

Briefly describe the 911/E911 capabilities of your solution.

The System i IP Telephony solution supports emergency services for users at any location on the corporate network, even as they move within the enterprise. An Emergency Response Locator (ERL) is defined as a location to which an emergency team may be dispatched. Each phone (uniquely identified by the assigned IP address) is part of a unique ERL and each ERL is assigned a location-wide emergency callback phone number. When a user’s IP phone is logged in at a different remote location, the IP phone will become a member of the location’s ERL, ensuring that emergency services are provided by the local media gateway. In addition, a set of emergency gateways are specified for an ERL, which are used to directly reach the emergency service provider in case the call processor(s) is rendered out of service.

Administrators can define emergency digits for each ERL in the system, which identifies the patterns defined for emergency service. Any numbers configured in the Emergency Digits section are marked as emergency calls and handled differently in case of a call disconnect. The 3Com IP Telephony solution supports the ability to define ELIN (Emergency Location Identification Number) as an identification number that gets assigned for a particular Emergency Call on a per-ERL basis. The ELIN is presented as the caller ID to the emergency services operator for an emergency call. The ELIN for a particular emergency call is chosen from a pool of available (Not In Use) ELINs. If a phone belonging to a particular ERL tries to make an emergency call, an ELIN that is not in use will be assigned and will be presented as the Caller ID to the emergency services operator.

Is operator intervention required or are specific phones required when dialing emergency calls?

No.



Describe how your system supports E911 services in conjunction with the local telephone operating company.

The 3Com VCX solution supports the ability to define ELIN (Emergency Location Identification Number) as an identification number that gets assigned for a particular Emergency Call on a per-ERL basis.

The ELIN will be presented as the caller ID to the Emergency Services Operator for an emergency call. Since this number is used as the callback phone number – this should be configured as a valid DID number. The ELIN for a particular emergency call is chosen from a pool of available (Not In Use) ELINs. If a Phone belonging to a particular ERL tries to make an emergency call, an ELIN that is not in use will be assigned and will be presented as the Caller ID to the emergency services operator.

If all the configured ELINs are currently in use, then the ERL’s Emergency Callback Phone number will be used as the Caller ID.

If the emergency services operator dials the ELIN (that was presented as the callerID), after the call reaches the VCX, the Authentication Server tries to associate the ELIN with an IP address based on the entries in the Emergency Contacts table. If an appropriate entry is found, then the call is sent to that IP address.

The emergency callback phone number associated with an ERL will be used as the caller ID for emergency calls when all the configured ELIN’s are currently in use.

How are media gateways assigned to handle emergency calls when the call processor is unavailable?

The VCX supports the ability for VCX administrators to define particular media gateways on a per-ERL basis to handle emergency calls in the case where both call processors are unavailable to route an emergency call.

The Emergency Gateway IP address specifies the address of the gateway to be used to reach an emergency service provider in case the VCX call processor(s) is rendered out of service. Multiple gateways can be configured for a redundant configuration.



This feature should not be confused with Remote Survivability. Remote Survivability provides an option to dial patterns, configured through DHCP option 184, directly through the gateway. Without Remote Survivability configured, only the patterns configured as Emergency Digits (specific SIP URI’s) can be dialed through the gateway.

When adding an emergency gateway IP, enter the IP address of the gateway for the ERL. One or more emergency gateway IP addresses can be specified per ERL.



2.9 Calling Features

Does the proposed solution support abbreviated dialing?

The 3Com VCX supports the Abbreviated Dialing feature. Abbreviated dialing allows phone numbers that are frequently used to be defined with a 4 or 5 digit extension.

Through a flexible dial plan, various site-to-site dialing plans can be implemented. The VCX can support either a 4 or 5-digit extension range. All users assigned to the dial plan have access to the abbreviated dialing pattern.

Does the proposed solution support anonymous call reject for next call and all calls?

The Anonymous Call Reject feature is supported by the VCX IP Telephony module. This feature gives the user the ability to block anonymous incoming calls.

Administrators can enable or disable this feature on a per-extension basis using the VCX Administrator web provisioning interface.

Users can enable or disable this feature on a per-extension basis using the VCX VoIP User web provisioning interface, as shown in the following example screen shot of the Call Restrictions feature.



Can the reception/operator console be disabled and have calls flow automatically to an alternate location?

The 3Com VCX solution supports the ability for users (and administrators) to define their call coverage point using the VCX VoIP User web provisioning interface in the current release. Call coverage defines how calls are handled when “all else fails”.

There are three things that can happen for call coverage:

Go to voice mail Go to an auto attendant Go to another phone number

o The VCX prevents loops on call forwarding

The default VCX coverage point is voice mail for each configured extension. When handling calls, the VCX examines call forwarding configuration first, then applies the user-set coverage. The VCX also supports “fallback” of calls back to coverage, which is enabled by default. If a user forwards their phone to a destination, and a call comes in but the forwarded station does not answer, then the VCX automatically falls back to the coverage point of the user who forwarded their calls.

The kinds of situations that encompass “all else fails” include:

If call forward universal is not set and the user is either not picking up or otherwise unavailable

User logged out Call forward on busy Call forward on ring/no-answer Any other call failure

An example of the VCX VoIP User web provisioning screen to configure call coverage is shown below.



Does the proposed solution support automatic answerback (hands free) mode?

Incoming calls will be auto answered in the hands free mode and the call will be answered on the speaker phone after an audible beep.

3Com’s 3102 Business IP phones have “hands free” buttons on them, and for any phone, users press an administrator and user-mappable button, use feature code 100, or select from the phone feature list. When the red light is on next to the “hands free” button, the phone is in hands free mode. Pressing the button again takes the phone out of hands free mode.

Can a call be placed to an extension and left until the extension becomes available, without altering forward on busy settings (camp on busy)?

The VCX IP Telephony module supports the Camp On Busy (also known as Automatic Call Back) feature.

Camp-On is a feature that is provided in two flavors. In both cases, the intent is for a caller to wait or be notified when a destination becomes available.

A camp-on invoked as part of a transfer causes the transferred party to wait on hold for the transfer destination to become available. If the transfer destination becomes available within a predefined time, the transferred party is connected to the transfer destination. The predefined time is referred to as the camp-on return interval (range: 30 to 300 seconds, default is 150 seconds).

If the predefined time elapses without the transfer destination becoming available, the transferred party is reconnected with the transferring party. To invoke the feature, while in a call the user would push the feature button and enter the feature code for camp-on followed by the destination to transfer to.

A camp-on invoked as part of a new call allows the caller to be freed of the call if the called party is unavailable and for that caller to be automatically called back and connected to the called party once the called party DOES become available. To invoke the feature, the user presses an administrator and user-mappable button, or uses feature code 469, followed by the destination to call.

An automatic callback timeout value is associated with this type of camp-on. If the called party does not become available before the automatic callback timeout expires, the camp-on is canceled (range: 5 to 60 minutes, default is 30 minutes).



Additionally:

More than one user could be camped on the same destination. In this case, waiting parties are serviced in the order they were camped.

Camp-on will be restricted by TOS settings. COS settings for this feature will be the same as those applied for a transfer or a call between the same parties.

Camp-on will work across sites. Camp-on will not work on PSTN numbers – only system extensions.

Feature Interactions:

Camp-on to a destination with DND or CFU will cause the camp-on to be queued until the appropriate timeout, or until the destination changes to an available state.

Camp-on to a hunt group, pickup group, and paging group is not allowed.

Does the proposed solution support Automatic Number Identification (ANI), Caller ID, Incoming calls and Privacy?

The VCX IP Telephony module supports the ability to display the calling party identification of an incoming call, when such calling party information is available.

If the appropriate Caller ID information is passed to the 3Com VCX IP Telephony call processor, it will be displayed on the phone (for all 3Com IP phones with displays). This includes all calls originating from internal or external phones.

Does the proposed solution support the ability to bridge multiple phones to a single phone extension?

Multi-site bridged call appearances are supported by the VCX IP Telephony module. Bridged call appearances allows the same phone number to appear and be answered on multiple phone sets.

If you bridge your extension to another one, you give permission to the owner of the other extension to add your extension to a button on their phone. Each user can give permission for up to 4 other extensions to add that user to a button on the other phone. The number of extensions that can actually be bridged is determined by the maximum number of contacts that the VCX system administrator has set up for the phone, which may be less than four.



The Bridged Call Appearance feature is implemented in the VCX call processor instead of the phone, providing improved functionality and feature interactions:

One to one line mapping between deviceso Map a Primary phone to one or more Secondary phoneso Secondary phones treat primary phone’s incoming calls as if call came on

their phone Everyone on the bridge can see the status of the line

o solid -> line in useo blinking -> incoming callo flashing -> call on hold

Shared Hold so that anyone can take the call when it is on hold Secondary line cannot be used to make outgoing calls, but merely as a speed dial to

the primary in case the line is free. Secondary phones (with permission) are able to see the MWI and retrieve mailbox

contents for the bridged extension. Multi-site bridged appearances are supported For call waiting, the phone is considered busy when all bridged extensions (except

for an originator in the bridged extension) are in use. Bridged extensions display caller ID information on 3Com phone displays Calls to bridged extensions go to the Primary’s coverage point when not answered

Users configure their Bridged Call Permissions and Bridge Mappings using the VCX VoIP User web provisioning interface. The following screen shot is an example of the VCX VoIP User web provisioning interface for Bridge Permissions. This screen shot illustrates extension 1709 giving permission for extensions’ 1702, 1703, 1704, and 1705 to map one of their Bridged Appearance buttons to extension 1709.

Using a 3Com 3102 Business IP phone set, there are up to 2 buttons (button numbers 4 and 5) that can be used for bridged appearances. If someone has bridged an extension to yours, you can map button 4 or 5 on your phone to the other extension.

The following screen shot is an example of the VCX VoIP User web provisioning interface for Bridge Mappings. This screen shot illustrates the mapping of extension 1709 to extension 1702’s button 4.



After this is done, a call to the other extension also rings on your telephone. To answer the call, pick up the handset and press the button to which you mapped the extension. The phone set that answers first will take the call, and the BLF lamp will only be displayed on the phone set that answered the call.

The bridged appearances are associated with up to 2 administrator and user-mappable buttons (depending on phone model), or using feature code 303.

Does the proposed solution support multiple appearances of the same extension (Automatic Line Selection). System must allow the user to automatically answer a predetermined line by lifting the handset.

The VCX allows the user to automatically answer a predetermined line by lifting the handset. For each phone, there is a maximum of 3 non-bridged appearances and 2 bridged appearances, providing a maximum of up to 5 call appearances allowed per extension. This is configurable by administrators on a per-phone basis.

The number of call appearances that are actually available depend on the phone being used. The 3Com 3100 Entry IP phone supports 1 line appearance. The 3Com 3101 Basic IP phones support 2 non-bridged call appearances, with up to 2 bridged appearances, providing up to 4 call appearances allowed per extension. The 3Com 3102 Business and 3103 Manager IP phones support up to 3 non-bridged call appearances, with up to 2 bridged appearances, providing up to 5 total call appearances allowed per extension.

The system appearances are associated with up to 3 administrator-mappable buttons (depending on phone model), or using feature code 200.

Does the proposed solution support three way calling native to the system?

Three Way Call Conferencing is a feature supported by the VCX IP Telephony module. This feature establishes an audio path (unicast) for multiple parties (up to 3) on a single call, established just via user keystrokes with no outside intervention.

The VCX 3-way call conferencing feature supports both unannounced and announced conferences:

Unannounced conference



o A user conferences in another person without notifying that person While on a call, press Conference (or feature code). The VCX places

the caller on hold. Dial the number of the person to be conferenced in. Press Conference (or feature code) again. The 3-way conference

begins when the person answers the call. Announced conference

o A user conferences in another person by first announcing the conference, giving the person the ability to decide whether to take the call.

While on a call, press Conference (or feature code). The VCX places the caller on hold.

Dial the number of the person to be conferenced in, then press OK. This establishes a one-way call when the person answers.

When the person answers, announce the conference. If the person accepts the conference, press Conference (or

feature code). Now three people are on the call. If the person does not accept the conference, hang up the

second call and go back to the first call by pressing the appropriate System Appearance button.

Three party conferencing is accessible via a fixed button on 3Com 3102 and 3103 IP phones, or using feature code 430.

Does the proposed solution support up to six party conferencing native to the system without any additional cost or equipment?

Six party conferencing is available for all IP Telephony users using 3Com Phones. Call reports will be generated when six party conferencing is used.

Useful for quick ad hoc conferencing and available at no additional cost. Daisy chaining allowed to enable occasional ad hoc conferences with more than 6

parties without purchasing full conference solution

System must allow a station user to define their call coverage point as voice mail, an auto attendant, or an internal/external phone number.

Enhanced call coverage for time of day, day of week and calendaro Time of day, day of week and calendar based call coverage is now available

for all extensions in addition to the coverage functionality in 7.0o More flexibility in defining coverage for all the userso Hunt groups also get coverage based on day, time, and calendar.

Does the proposed solution support call drop?

The VCX IP Telephony module supports the Call Drop feature. Call Drop allows a user to terminate a call without hanging up the receiver.

All of the 3Com IP phone sets include a “Release” button, which will terminate the call without hanging up the receiver. The Release button is accessed via an administrator and user-mappable button (depending on phone model), or using feature code 111).



Does the proposed solution support call forward busy?

The VCX IP Telephony module supports the Call Forward Busy feature. The Call Forward Busy feature allows a user to redirect calls to another station or location when their extension is busy.

Users can configure Call Forward Busy on 3Com IP phone sets by pressing an administrator and user-mappable button, selects from the phone feature list, or using feature code 467. Users can also configure Call Forward Busy using the VCX VoIP User web provisioning interface.

Does the proposed solution support call forward all (call forward universal)?

The VCX IP Telephony module supports the Call Forward All feature. The Call Forward All feature allows a user to direct all calls to another station or location.

Users can configure Call Forward All on 3Com IP phone sets by pressing an administrator and user-mappable button, selects from the phone feature list, or using feature code 465. Users can also configure Call Forward All using the VCX VoIP User web provisioning

Does the proposed solution support call forward - no answer?

The VCX IP Telephony module supports the Call Forward No Answer feature. The Call Forward No Answer feature allows a user to redirect calls to another station or location when their extension rings with no answer.

Users can configure Call Forward No Answer on 3Com IP phone sets by pressing an administrator and user-mappable button, selects from the phone feature list, or using feature code 466. Users can also configure Call Forward No Answer using the VCX VoIP User web provisioning interface.

Does the proposed solution support the ability to provide an audible tone to remind the user that their station is in the call forward mode?

Provides telephone users with a visual display of the forwarding phone on the LCD and an audible tone to remind the user that their station is in the call forward mode.

Does the proposed solution support the ability to remotely forward an extension?

Remote (third party) call forwarding is supported by the VCX solution. The remote call forwarding feature allows other users on the system to forward your extension to another number.

VCX administrators provision an Access Control List (ACL) to define the other users who have permission to perform the remote forwarding for any given extension. This is a multi-site feature, in which any user at any site can perform the remote call forward for you.



For example, this feature is used when the VCX VoIP User provisioning interface is not available for a user who wishes to have their phone forwarded to another number. The user simply contacts one of the users on their Access Control List, and informs them of the number to forward their phone to. The user doing the remote call forwarding uses an administrator and user-mappable button, uses feature code 468, or selects from the phone feature list to set the call forwarding or to un-forward the phone.

The number set for forwarding is restricted according to existing controls on call forwarding for the user being forwarded.

When enabled, the remote (destination’s) phone LED (if available) flashes for all forwarded calls, and the display (if available) shows a call forward message.

Does the proposed solution support the ability to restrict call forward to trunk?

The 3Com VCX solution supports the ability to restrict call forwarding to trunks. This can be toggle on and off by system administrators on a class of service and user basis.

When enabled, this feature prevents users from forwarding their station to an external phone number.

Does the proposed solution support a programmable one-button send all calls?

The 3Com VCX solution supports a “one-button send all calls” feature natively on 3Com IP Business phone sets.

A user simply presses the “Call Forward” button and then specifies what number the calls should be forwarded to. This can include internal extensions or outside numbers if their class of service allows this.

Does the proposed solution support call history with inbound/outbound/missed calls?

The VCX IP Telephony module supports Call History logs in the current release. Call History logs are available on all 3Com IP phone sets that have displays, and from the VCX VoIP User web provisioning interface.



The Call History feature can be used to display a phone’s call logs. These are the logs of the 10 most recent calls to and from the phone set. From the Call History users can select calls and the phone automatically dials them.

Each 3Com phone set with a display includes 3 different Call History log files:

Placed calls Received calls Missed calls

The Call History feature is invoked by pressing an administrator and user-mappable button, uses feature code 462, or selects from the phone feature list. The Scroll buttons on the phone set are used to navigate through the list and select one of the options:

Placed calls (press 1) Received calls (press 2) Missed calls (press 3) Unreviewed missed calls (press 4) Clear all call logs (press 5)

The display panel always starts with the oldest call in the selected category. That is, the oldest call appears first and the most recent call appears last. The display panel scrolls through the calls one at a time. After the last call, this message appears in the display panel for Placed and Received calls:

No more history

This message appears for missed calls:

No more missed calls

The Soft buttons have the following functions when viewing the Call History:

To select a call from the list and dial the call automatically, press the Slct button (left-most)

To return to the previous menu, press the Back button (center) To exit the Call History display, press the Exit button (right-most)

In addition, the Call History of each phone can be viewed from the VCX VoIP User web provisioning interface.



Does the proposed solution support call hold?

The VCX IP Telephony module supports the Call Hold feature. All of the 310x series of 3Com IP phone sets include a “Hold” button, which is a fixed button (hard key) on the phone. The VCX solution also supports the ability to remind users of calls on hold.

To put a call on hold, the user presses the “Hold” button, or uses feature code 402. The user’s phone displays a “Hold” message and the call appearance lamp blinks. The user is reminded of the held call with the blinking appearance lamp and an audible beep after a configurable amount of time.

If configured by the VCX administrator, the caller will hear Music On Hold audio. The particular Music On Hold the callers will hear is configured on a per-user basis by administrators.

A user can put a call on hold, dial a new call, and toggle between the two calls. A user can also put a current call on hold, answer a second call, and then toggle between the two calls.

Does the proposed solution support a call hold reminder?

The user is reminded of the held call with the blinking appearance lamp and an audible beep after a configurable amount of time.

Does the proposed solution support the ability to place a call in a parked state, similar to hold, where it can be retrieved by any attendant console or by another telephone?

The VCX IP Telephony module supports the Call Park feature. The Call Park feature is used to place a call in a holding pattern and make it available for you or for another user to pickup from any phone set on the system.

The Call Park feature is useful when the recipient is elsewhere in the building or you want to continue a call on another phone set and transferring the call does not give you enough time to retrieve it.

The Call Park feature can be used on 3Com IP phone sets via an administrator and user-mappable button, or uses feature code 444. When you park a call, you assign it a call park



extension, which you or someone else uses to retrieve it. The factory default call park extension numbers are 800 through 899 inclusive.

When a call is successfully parked, the parking party’s line is automatically freed. The caller who is parked will hear the Music On Hold music configured for the user who initiated the call park. The call can be retrieved from any 3Com IP phone set by dialing the call park extension.

If a call is left parked beyond the configurable call park timeout period (default is 5 minutes), the user who parked the call will be called back. If the user who parked the call is unavailable, the call will be sent to the call coverage point of the user who parked the call.

The Call Park feature is implemented in the VCX call processor instead of the phone, providing improved functionality and feature interactions:

There is no limit to the number of calls that can be parked by a user. In previous VCX versions, each user was limited to parking 3 calls because the line was not automatically freed, using up their 3 available system appearances. The number of calls that can be parked system wide is limited only by the configured call park range.

If the specified park number is in use or if no park number is selected, the VCX call processor will automatically select the next available park number. In previous VCX versions, this resulted in a busy condition forcing the call park initiator to select a different call park extension.

VCX Call Park functionality is supported for all endpoints, including FXS ports.

To park a call:

While you are on a call, press the “Call Park” button or feature code 444. The display panel shows a default Call Park extension. The caller hears the Music On Hold configured for the user who parked the call.

Press the Call Park button (or feature code) again to park the call on the displayed default Call Park extension, or enter a different Call Park extension and press ok.

o If a selected Call Park extension is already in use, the display panel shows:Park Number Rejected. Reenter number.

o Try another Call Park extension

To notify another user about the parked call:

From a 3Com IP phone set, select an available System Appearance line and dial the user’s extension.

When the call is answered, tell the user the Call Park extension number. Hang up The user dials the Call Park extension and the VCX connects the call automatically.

To retrieve a parked call yourself:

Pick up the handset of any 3Com IP phone set on the system.



Dial the Call Park extension that was assigned to the call. The VCX connects you to the parked call.

Does the proposed solution support the ability to fall back to a parked call?

The VCX IP Telephony module supports the Call Park Fallback feature.

If a call is left parked beyond the configurable call park timeout period (default is 5 minutes), the user who parked the call will be called back. If the user who parked the call is unavailable, the call will be sent to the call coverage point of the user who parked the call.

Does the proposed solution support directed call pickup?

The 3Com VCX IP Telephony solution supports the Directed Call Pickup feature. The Directed Call Pickup feature allows a user to pickup a call ringing on their phone on another phone by invoking the Directed Call Pickup feature and entering an authorization code followed by the extension to be picked up.

By enabling this feature and assigning a security code, the Directed Call Pickup feature allows a user to answer their own phone from another desk. To answer the call, a person presses an administrator and user-mappable button, or uses feature code 455 from any telephone in the network, enters their security code, then enters the extension of the ringing phone. This transfers the call to the phone they are on.

For example:

455 is the feature code for call pick up 52 is the security code 1702 is the phone you want to pick up

So, when user Bob (1701) wants to pick up a call that is ringing on 1702, Bob would dial Feature + 455, 52#, 1702. This will pick up the call being placed to 1702.

The Directed Call Pickup feature is implemented in the VCX call processor instead of the phone, providing improved functionality and feature interactions. With its standard SIP implementation, the VCX Directed Call Pickup functionality is supported for all endpoints, including FXS ports and non-3Com SIP phones.

System must allow a group of telephones to answer a ringing station in its group through the use of either an access code or a programmed pickup button.

The Call Pickup Group feature is supported by the VCX IP Telephony module.

The Call Pickup Group feature allows a user to answer a call ringing on another extension where both extensions are part of a pickup group. Callers can pick ringing calls from among a group of phones. Calls are picked by oldest ringing call within the group. There can be up to 50 members per call pickup group and up to 800 call pickup groups per site (call processor).



To invoke the Call Pickup Group feature, the picking party simply dials the number of the pickup group. If there are pending calls to the pickup group, the picking will be connected to the oldest ringing call. The ringing destination will have the call canceled (will stop ringing). If there are no pending calls, the phone will receive a failure code from the Call Processor that indicates there are no available calls (same as directed call pickup).

To configure the feature, an administrator creates a new group of type Pickup Group. The Pickup Group has the following properties:

Name Number Member List Allow Anyone To Use Group (checkbox/Boolean)

Additionally:

Optionally, the pickup group can be configured to allow phones that are not members of the pickup group to use the group to answer calls. This is a simple global flag and not controlled via access list.

If a group is configured to allow non-members to answer group calls, a user from another site would be allowed to pick a call – however there will be no specific notification sent to those remote phones that there are calls pending in the group.

Pickup group members will be restricted to one site. COS will not apply to pickup calls; if the picking party is a member of the group (or if

everyone is allowed to use the group) then the operation will succeed. Similarly there will be no additional setting for group pickup in regards to TOS. Access to this feature will be controlled only through group configuration.

Feature Interactions:

Hunt group calls can be picked if the hunt group member is also a member of the pickup group.

A pickup group cannot be used to pick-up a bridged call. Call Forwarding: If the forwarding station and the forwarded-to station belong to the

same Call Pickup Group – then Call Pickup can be used to retrieve the ringing call Group pickup cannot pick up a camp-on callback



Does the proposed solution support call restrictions for blocking inbound, blocking outbound, white list, and black list?

The Call Restrictions (Blocking Inbound) feature is supported by the VCX IP Telephony module. The Call Restrictions feature allows users to selectively block calls from user-defined origins, including incoming calls, specific extensions, or specific calling party numbers.

This can be setup in the dial plan for system-wide blocking as well as white list/black list functionality on a user-by-user basis. Users configure Call Restrictions with the VCX VoIP User web provisioning interface, as shown in the following example screen shot of the Call Restrictions feature.

The Call Restrictions (Blocking Outbound) feature is supported by the VCX IP Telephony module. The Call Restrictions feature allows users to selectively block calls, including outgoing calls.

This can be setup in the dial plan for system-wide blocking as well as white list/black list functionality on a user-by-user basis. Users configure Call Restrictions with the VCX VoIP User web provisioning interface.

The Call Restrictions (Call Screening) feature is supported by the VCX IP Telephony module. The Call Restrictions feature allows users to selectively block calls, including outgoing calls.

This can be setup in the dial plan for system-wide blocking as well as white list/black list functionality on a user-by-user basis. Users configure Call Restrictions with the VCX VoIP User web provisioning interface.

The VCX IP Telephony module supports White List functionality. This provides a “permit” list for incoming calls.

The VCX IP Telephony module supports Black List functionality. This provides the ability to block incoming calls based on a specific number or pattern.



Does the proposed solution support call restrictions for toll screening?

The VCX IP Telephony module supports Toll Screening functionality. This is a class of service feature that provides an access list to restrict the user’s ability to make toll calls.

Does the proposed solution support call return?

The VCX IP Telephony module supports the Call Return feature. The Call Return feature allows a user to call back a previous incoming call by the press of a button.

On 3Com IP phone sets with displays, the Call History button can be used to review the most recently received calls. The left-most button under the display can be pressed to automatically dial the originator of a call using the provided calling party information.

Describe how the proposed solution supports attended (supervised) call transfer?

The VCX IP Telephony module supports the Attended Call Transfer feature. Also known as a Supervised Call Transfer, an Attended Call Transfer allows a user to transfer a call by announcing it to the recipient.

While on a call, a user presses the “Transfer” button or feature code 420, which places the caller on hold. The caller will hear the Music On Hold music configured for the user who initiated the transfer. The user dials the number to which they want to transfer the call, then presses the “Transfer” button again or a feature code.

For an Attended Call Transfer, the user stays on the line and waits for the recipient to answer. If the recipient answers, the user announces the call to the recipient. If the recipient wants to take the call, the user presses the “Transfer” button or a feature code. The caller will be transferred to the recipient.

If the recipient does not want to take the call, the user hangs up on the call with the recipient, and returns to the original call.

The Music On Hold audio originates from the 3Com IP Messaging module. If the IP Messaging module is not present or the user initiating the transfer is not configured for MOH, the user being transferred will hear silence until connected.

The Call Transfer (Attended) feature is implemented in the VCX call processor instead of the phone, providing improved functionality and feature interactions.

Describe how the proposed solution supports recovery of mis-operation of transferred calls, which prevents external calls from being dropped due to a station user’s incorrect operation of transfer feature.

The 3Com VCX solution prevents callers from being dropped due to transfer failures by transferring the call back to the station which initiated the transfer.

Describe how the proposed solution can restrict call transfers to an outgoing trunk?



The 3Com VCX solution can restrict call transfers by phones and users based on class of service settings and black/white lists.

Describe how the proposed solution supports unattended (Blind) call transfers.

The VCX IP Telephony module supports the Unattended Call Transfer feature. Also known as a Blind Call Transfer, an Unattended Call Transfer allows a user to transfer a call without notifying the recipient.

While on a call, a user presses the “Transfer” button or feature code 420, which places the caller on hold. The caller will hear the Music On Hold music configured for the user who initiated the transfer during the time the initiator is dialing the transfer number. The user dials the number to which they want to transfer the call, then presses the “Transfer” button again or a feature code.

Once the transfer is initiated, the VCX disconnects the line between the caller and the user who initiated the transfer. This frees up the line without requiring the user to disconnect the call. The caller now hears ring back until the call is answered.

After starting an unattended transfer, if the transfer cannot be completed due to busy/ring no answer/wrong number/DND/logged out/etc., the user who initiated the transfer is called back. If the transfer initiator cannot be contacted, the caller will be sent to the coverage point for the transfer initiator.

The caller will be transferred to the recipient, and will encounter the treatment defined for the recipient’s phone upon ring/no answer or busy.

The Music On Hold audio originates from the 3Com IP Messaging module. If the IP Messaging module is not present or the user initiating the transfer is not configured for MOH, the user being transferred will hear silence until connected.

The Call Transfer (Unattended) feature is implemented in the VCX call processor instead of the phone, providing improved functionality and feature interactions. VCX Call Transfer functionality is supported for all endpoints, including FXS ports.

Describe how the proposed solution supports call waiting?

The 3Com VCX IP Telephony module supports the Call Waiting feature. The Call Waiting feature provides a “beep” on a current call to inform the user that another call has arrived on another access line.

When the “beep” is heard, the user puts the current call on hold by pressing the “Hold” button or a feature code, then presses a System Appearance (or Toggle) button to connect with the new call. To toggle between the two calls, put the current call on hold and toggle to the other call.



If available and up to the maximum allowed by the phone set, a System Appearance lamp will blink and 3Com IP phone sets with displays will present the calling party identification (if available) when another call arrives.

For all 3Com IP Phones on VCX system with release 6.0 and higher, users will hear a half ring (not a beep inband with audio) just like the NBX phones.

Users can now configure their telephones to flash instead of playing a waiting tone when calls arrive while the users are on a call. The waiting tone can be configured as a single, double, or triple tone. “No ring” is also available.

Can the proposed solution allow a user to override a COS block which may be tied to an extension? For example, can an authorized user override the international COS restrictions on a phone in a conference room?

The 3Com VCX IP Telephony module supports the ability for a user to override a class of service (COS) restriction that may be defined for a particular extension.

This feature allows a user to invoke a Class of Service Override by pressing an administrator and user-mappable button, using feature code 433, or selecting from the phone feature list. The user then enters their extension and password, which are recorded in a CDR. This logs a user into their extension on any phone in CoS Override mode, but only for the duration of one call.

After logging in using this method, the user can place one call using their class of service permissions. At the completion of the call, regardless of call termination status, the phone automatically reverts to the previous extension’s class of service definition.

Describe the support for direct inward dialing (DID).

The 3Com VCX IP Telephony solution supports Direct Inward Dialing (DID) on the digital media gateways. DID connects calls from the PSTN directly to a dialed extension number without attendant assistance. Specialized DID trunk circuits from the service provider are required to implement this feature.

Describe the support for direct outward dialing (DOD).

The 3Com VCX IP Telephony module supports Direct Outward Dialing (DOD). The DOD feature allows users to access the PSTN without attendant assistance.

The flexibility of the VCX dial plan configuration allows <<clientShort>> to define how users will dial DOD numbers, such as 9+.

Describe how the proposed solution supports distinctive ring patterns for different types of calls.

The VCX IP Telephony module supports the ability for users to configure distinctive ring patterns for different types of calls, including:



A call from within your organization An outside call A private call

The VCX VoIP User web provisioning interface is used to configure the distinctive ring patterns for internal, external, and private calls. There are 9 different ring tones that can be configured each with one, two, or three rings. The VCX VoIP User web provisioning interface allows the user to hear the ring pattern before selecting by using a standard web-based media plugin (not associated with VCX, but part of standard user PCs).

The following screen shot is an example of the VCX VoIP Users web provisioning interface Ring Patterns feature.

Describe how the proposed solution supports distinctive ring patterns for different phone numbers.

The VCX IP Telephony module supports the ability for users to configure distinctive ring patterns for different phone numbers.

The VCX Selective Ringing feature is used to choose the tone to hear when an incoming call from a specific telephone number is received. Up to 10 telephone numbers (internal or external) can be configured, each with its own distinctive ring pattern.

The VCX VoIP User web provisioning interface is used to configure the distinctive ring patterns for specific phone numbers. There are 9 different ring tones that can be configured each with one, two, or three rings. The VCX VoIP User web provisioning interface allows the user to hear the ring pattern before selecting by using a standard web-based media plugin (not associated with VCX, but part of standard user PCs).

The following screen shot is an example of the VCX VoIP User web provisioning interface for the Selective Ringing feature.



Does the proposed solution support DNIS (Dialed Number Identification Service)?

The 3Com VCX IP Telephony solution supports DNIS. The VCX IP Telephony module receives the DNIS (called number) in the SIP Invite message from a VCX media gateway, which receives the DNIS from the adjacent switch using the configured signaling method.

System must allow the station user or attendant to place their station in the “Do Not Disturb” mode.

The 3Com VCX IP Telephony module supports the Do Not Disturb feature. The Do Not Disturb feature is used to route all incoming calls to the call coverage point defined for the phone.

The 3Com IP phone sets provide Do Not Disturb (DND) enable/disable capabilities using an administrator and user-mappable button, using feature code 446, or selecting from the phone feature list.

When a phone is in Do Not Disturb mode:

An incoming call does not cause your phone to ring You can use the phone to dial outgoing calls

Do Not Disturb is a call processor-based feature. This allows the Do Not Disturb feature to be invoked from any device that can implement feature codes. The Do Not Disturb feature is uniform across all appearances of a phone number. If multiple phones are logged in using the same phone number, and one of those phones invokes DND, all phones will have DND turned on.

If Do Not Disturb is turned on with one or more pending calls ringing, all of those calls will be sent to coverage (except for hunt group calls and bridged calls for secondary users of the bridge). After this point, DND will be turned on for all subsequent calls.

System must support Dual Tone Multi-Frequency (DTMF) end-to-end signaling through an established outgoing connection.

The 3Com solution supports end-to-end DTMF signaling via SIP RFC 2833 for incoming and outgoing calls.



Describe how the proposed solution supports feature codes.

The feature codes allow all of the features to be used from any 3Com SIP phone or an analog phone connected to an FXS media gateway. The feature codes between NBX and VCX are the same.

Phone-related feature codes are identified in the following table.

Feature CodeAdmin phone

mappableUser phone mappable

Number Of Parameters

HANDS FREE TOGGLE 100 X X 0

MUTE BUTTON 101VOLUME UP BUTTON 102VOLUME DOWN BUTTON 103SPEAKER TOGGLE 104SOFT1 BUTTON 105SOFT2 BUTTON 106SOFT3 BUTTON 107LCD UP BUTTON 108LCD DOWN BUTTON 109Feature 110 X X 0

Release 111 X X 0

OK 120 X X 0ALPHA NUMERIC TOGGLE 122LCD LEFT BUTTON 154LCD RIGHT BUTTON 155DISPLAY IP ADDRESS 320DISPLAY GATEWAY IP ADDRESS 321DISPLAY SUBNET MASK 322DISPLAY NCP IP ADDRESS 323DISPLAY NCP2 IP ADDRESS 324DISPLAY CURRENT PBX IP 325DISPLAY SIP GATEWAY INFO 326DISPLAY SUBJECT INFO 327DTMF VOLUME 329CONFIG BEEP 330SEND BEEP (beep feature is a page sent between phones – your phone tells you who beeped you) 331



Feature CodeAdmin phone

mappableUser phone mappable


TOGGLE WARNING TONE(busy tone generated when phone is off hook past a certain time – like a pstn phone) 332RESET PHONE TO DEFAULTS 333FORCE REGISTER 335SOFT RESET 336HARD RESET 337DISPLAY DEBUG INFO 338TOGGLE ASSERT BEHAVIOR 339USE 20MS FRAME SIZE 340USE COMPRESSED CODEC 341USE 729B 342USE SUBNET OPTION 343DIAL BY URL 344HELP 345 X X 0

REDIAL BUTTON 401HOLD BUTTON 402PROGRAM BUTTON 410TRANSFER BUTTON 420CONFERENCE BUTTON 430CLASS OF SERVICE OVERRIDE 433

X X 0

USER PASSWORD (set locally on phone only) 434USER DIRECTORY 461 X X 0

CALL HISTORY 462 X X 0

View Personal Speed Dial 463 X X

View System Speed Dial 464 X X

VERSION INFO 837VIEW HUNT GROUP 972 X X 0

Phone Login/Logout 128 X X 0

Global User Directory 129 X X 0

System Appearance 200 X 0

Bridged Appearance 303X 1 (bridged

number)Headset Enable/Disable 112 X X 0

System-related feature codes are identified in the following table.

Feature Code Admin phone

mappable

User phone

mappable


Feature Request String

Malicious Call Trace 119 0 *fcSilent Monitor 425 X 1 *fc*monitor_num



Feature Code Admin phone

mappable

User phone

mappable


Feature Request String

Barge In 428 1 *fc*barge_in_numFwdMail Toggle 440 X X 0 *fcTransfer To Voicemail (Coverage)

441 X X 1 *fc*destination

Park 444 X X 1 *fc*park_numDo Not Disturb 446 X X 0 *fcDirected Pickup 455 X X 2 *fc*pwd*pickup_numConfig Forward Universal 465 X X 0 in b-map

1*fc*fwd_number

Config Forward Busy 467 X X 0 in b-map1

*fc*fwd_number

Config Forward Ring No Answer

466 X X 0 in b-map1

*fc*fwd_number

Config RemoteFwdUniversal 468 X X 2 *fc*src_num*fwd_numCampOn 469 X X 1 *fc*destination NumberAttendant Serial Call 471 X X 1 *fc*transfer DestinationRetrieve Voice Mail 600 1 *fc*user (optional)

(*user could be another mailbox accessible to this phone)

Personal Speed Dial 601 X X 1 (invoke)2 (config)

*fc*speed dial index*number(*number is optional and used to config the speed dial through TUI)

System Speed Dial 700 X X 1 *fc*speed dial indexBlock Caller Id Toggle 889 X 0 *fcBlock/Unblock CallId for Current Call

890 1 *fc*destination number

Hunt GroupToggle* 971 X 1 *fc*hunt_group_numBlock Monitor/Barge 429 X X 1 *fc*destinationCall 125 X X 1 *fc*destinationBusy Lamp Field 126 X 1 *fc*destinationConfig Button Map 127 2 minimum *fc*buttonNumber

*featureRequestString

Success codes for features are identified in the following table.

Code Description1000: Success General success case used in many features1001: Park Success: XXX Specific park success where the park number is returned

as well (XXX)1002: Do not disturb enabled Do not disturb feature enabled1003: Do not disturb disabled Do not disturb feature disabled1004: Hunt Group Login: XXX Hunt group login where hunt group number is XXX

1005: Hunt Group Logout: XXX Logout of hunt group where hunt group number is XXX

1006: Fwd to mail enabled Indicates to phone that fwd to mail is successfully enabled



1007: Fwd to mail disabled Indicates to phone that fwd to mail is successfully disabled

Failure codes for features are identified in the following table.

Code Description9999: Unauthorized User is unauthorized to perform the feature9998: No call for feature Mid call feature invoked outside of a call9997: Invalid within call New call feature invoked within a call9996: No available calls There are no calls available to perform the specified feature

on. This is used for features where a user is trying to manipulate a call on a remote user (silent-monitor, barge-in, pickup)

9995: Fwd number invalid Used in TUI based forward configuration to indicate that the forwarding number selected is not an allowed forwarding number for that user (Class of Service may restrict forwarding numbers)

9994: All park numbers busy Used in response to a park request to indicate that the entire range of park numbers for this callP is in use and no more parks can be performed until a number is freed up.

9993: Unknown user Used in response to features that specify a user as an argument (remote fwd Config, transfer to vmail, etc…) to indicate that the specified user is not known in the system.

9992: Unknown Feature Code Used in response to a feature code request with a code that is not known to the call processor.

9991: Unable to handle feature Feature request sent in wrong call state (in-call feature request sent without a call, or out-of-call feature request sent within the context of a call).

9990: Wrong number of params Feature request sent with wrong number of additional feature parameters in the feature request string.

9989: Server Error, profile Error on callP when attempting to communicate with the auth server to retrieve a user profile.

9988: Server Error, permit Error on callP when attempting to communicate with the auth server to check permissions on a call.

9987: Server Error, feature Error on callP when attempting to communicate with the auth server to invoke a feature operation.

9986: Server Error, no voice mail

Error on callP during a request to connect to voice mail, where callP cannot find a route to the users voice mail server.

Describe how the proposed solution supports ability to forward calls to voice mail.

The VCX IP Telephony solution supports the Forward to Mail feature. This is a call processor based feature and can be invoked by any device that can implement feature codes.

The Forward Mail feature is invoked using the Forward Mail button on 3Com 3102 IP phone sets, an administrator and user-mappable button, using feature code 440, or selecting from the phone feature list. The Forward to Mail feature is uniform across all appearances of a phone number. If multiple phones are logged in using the same phone number, and one of those phones invokes Forward to Mail, all phones will have Forward to Mail enabled.



When Forward to Mail is enabled, incoming calls will ring once on the destination user’s phone after which calls will be directed to the voice mailbox of the user. The one ring is a quick ring which is intended to allow the user to pick up if they wish to.

If Forward to Mail is enabled with one or more pending calls ringing, all of those calls will be sent to voice mail (except hunt group calls, and bridged calls for the secondary users of the bridge).

Describe how the proposed solution supports a global user directory.

An integrated global user directory is supported by the VCX IP Telephony module. The enterprise-wide VCX user directory is available on 3Com IP phone set displays and from the VCX VoIP User web provisioning interface. The directory provides location/site name, phone number, first name, and last name information.

Using the Directory button on a 3Com 3103 Manager IP phone, an administrator and user-mappable button, or using feature code 129, users can scroll through the global directory one entry at a time and select an entry for dialing. Search by name is available through key presses with this telephone user interface.

The VCX global directory is global across all locations. Specific users and their phones can be excluded from this directory. Administration of the VCX global directory is performed by VCX system administrators using the VCX IP Telephony Admin web provisioning interface. The VCX Data Servers maintain the global directory listing as an XML file.

After being started, the User Directory display panel shows the first user in the directory. The Scroll buttons are used to locate a particular user. The Soft buttons have the following functions when viewing the User Directory:

Use the Select button (left-most) to select a user and dial that user’s extension Use the Back button (center) to display sort order options:

o Press the Select button to sort by first nameo Press the Clear button to sort by last nameo Press the Exit button (right-most) to sort by extension

Use the Exit button to return to the default display panel

Users can also access the global VCX user directory when logged into a mailbox or calling into an auto attendant when using the IP Messaging module.

The following is an example screen shot of the VCX Directory from the VCX VoIP User web provisioning interface. Clicking on the “Search Global Directory” button will allow the user to search both the local and global directories.



Describe how the proposed solution supports the ability to automatically dial a specific number when a handset is picked up (hot ring down).

The 3Com VCX IP Telephony module supports the Hot Line feature. The Hot Line, or Hot Ring Down, feature allows a phone to connect to a pre-determined number as soon as the user picks up the phone.

This feature is used often in elevators, common hotel areas, etc.

Describe the hunt group functionality of your system, including interactions with other features and system capacities.

The VCX IP Telephony module supports comprehensive multi-site hunt group features. The VCX Hunt Group feature provides ease of administration and use, strong interaction with other features, and detailed reporting. VCX Hunt Groups can be implemented in several different configurations, providing a better experience for your callers and improving customer satisfaction. Hunt group members can be part of any branch and regional call processor but still part of an enterprise wide hunt group and support enhanced coverage.

A Hunt Group is a group of existing VCX users/extensions, which has a virtual extension. Calls to the hunt group virtual extension are queued and hunt group members are served following the selected algorithm. Members of the HG have the option to log in or out of the HG. If logged out, the HG won't try to call them.

The VCX supports viewing “in hunt group” queue indicators on the phones and supports per-hunt group thresholds based on time/number in queue. Delayed or No Ringing options are available for hunt groups.

The VCX supports the following types of hunt groups:

Linearo Single pass through list of members, bounce to coverage at end of list

Circularo Cyclic passes through list of members, bounce to coverage after pre-set

timeout Calling Groups

o Simultaneous ringing of all list members, bounce to coverage after pre-set timeout

In the current release, the VCX supports the following hunt group capacities:



A maximum of 75 members per Hunt Group is permitted. A maximum of 25 simultaneous hunt group login per member. A maximum of 512 calls can be queued per local system. A maximum of 100 Hunt groups allowed per local system.

The following table summarizes the behavior for the different hunt group types supported by the VCX.

HG Type

No Members

Members Logged Out Members Busy Total Timeout

Linear Go to Call Coverage

Go to Call Coverage Go to Call Coverage Go to Call Coverage

Circular Go to Call Coverage

Wait for somebody logs in till Time out

Wait for somebody to get free till Time out

Go to Call Coverage

Calling Go to Call Coverage

Go to Call Coverage Queue this call and wait for somebody to get free till Time out

Go to Call Coverage

The following are examples of the provisioning screens for an example hunt group.



Hunt Group Mailboxes and MWI

VCX hunt groups have their own call coverage path with multiple options and a unique voicemail box number. Hunt group members have the ability on their phone to monitor MWI and message status, and to login/connect to that mailbox and retrieve the contents.

This allows users to see the MWI and message status and retrieve contents for their own personal mailbox and the hunt group mailbox. Non LCD phone will directly connect to an IP Messaging auto attendant after pressing message button where the mailbox number will be asked. Hunt group members will get MWI and message status even if they are logged out of a hunt group.

Hunt Group Login and Logout

For logging in and out of hunt groups, the VCX supports both static and dynamic logins. For static logins, once a hunt group member is logged in, no future logins are required, even if the user logs out from their phone. For dynamic logins, hunt group members have the ability to login and logout of the hunt group as needed by pressing an administrator-mappable button, using feature code 971, selecting from a feature list, or from the VCX VoIP User web provisioning interface.

Hunt Group Phone Display

Hunt group members have the ability to view all hunt groups that the user is logged in, by pressing an administrator-mappable button, using feature code 972, selecting from a feature list, or from the VCX VoIP User web provisioning interface. Whenever the call rings on one of the hunt group stations, the caller ID will display that the call is for XXX Hunt Group (Name and number) and displays the callers’ name/number. Once the user picks up the call, the display will show calling party’s caller ID.



The VCX Hunt Group feature provides strong interaction with other VCX features, improving the experience for administrators, hunt group members, and callers. Hunt groups have the following interactions with other VCX features:

Hunt Group class of service (COS) will be applicable on all incoming calls to hunt group

Primary users class of service (COS) will be applicable on hunt group calls which are transferred to other destination.

A hunt group can be added to a conference. A call answered can be put on hold and taken off hold without losing the caller. If a hunt group member puts a call on hold, they can receive other calls. A Hunt group call on hold will hear hunt group MOH. A call answered from a Hunt Group can be parked and then picked up by any user

on the same site. If the 3Com Phone has Hands Free enabled, a call coming into a Hunt Group will not

be picked up automatically. User personal settings (Call Forward, coverage, DND etc.) are not invoked on Hunt

Group calls. An external call to a private user (someone not in the hunt group) can be transferred

to a Hunt Group. Both attended and unattended transfers are allowed for a call to be transferred to a

Hunt Group or from a hunt group. After reaching an Auto Attendant the user can enter the Hunt Group extension and

be transferred to a Hunt Group. Internal/external callers that are forwarded to voice mail can leave a message and

maneuver through the voice mail options. Calls are re-queued for Hunt Groups only when forward to voicemail (or AA) fails

because of non-availability of Voicemail ports Each time a call is put back in the Hunt Group queue, it is treated like a new call

Hunt Groups and Bridge Line Interaction

A primary or a secondary bridged phone can be part of a hunt group. If the primary is a member of a hunt group then a call coming from the hunt group will ring on primary’s System Appearance (SA) line. The secondary cannot take the primary’s hunt group call. The



primary phone should have at least one SA line if it needs to join a hunt group. If the primary has DND enabled, the call will still alert on the primary and the DND will be ignored.

Hunt Groups and Malicious Call Trace

Hunt group members who initiate a Malicious Call Trace (MCT) are automatically logged out of all hunt groups to optimize MCT call handling. Although logged out of the hunt group, the member telephone can still receive direct dialed extension calls.



Describe how the proposed solution supports a calling group hunt group.

The VCX solution supports single-site calling group hunt groups in the current release. A calling group hunt group is simultaneous ringing to all members of a list with timeout, and call coverage to a hunt group mailbox.

Calling group hunt groups are implemented as a special case of hunt group where a single call rings on all members of the hunt group. 310x the phones in a calling group continue to ring until either a member answers the call or total timeout elapses. The per device timeout is redundant for a calling group. A Calling Group call also calls on a member’s phone that is busy or on another call. In this case the call is treated like call waiting. A logged out member is not called.

Only one call is served out of the calling group queue, with the other calls waiting to be served (queued) or routed to call coverage after total timeout. The calling party will hear announcement/MOH while waiting in the calling group queue. If all members are logged out then the call is forwarded to call coverage immediately. If there are no members in the calling group, the call is forwarded to the call coverage path without waiting.

If two or more members of the calling group try to answer the same call at the same time, only one member will be connected to the call. The other member will hear dial tone. A maximum of 512 calls can be queued. If this limit is reached the new call will go to call coverage.


Ring...Ring...

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Ring...Ring...

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Ring...Ring...

Ring...Ring...

Ring...Ring...

Ring...Ring...

Ring...Ring...

Ring...Ring...

Ring...Ring...

Ring...Ring...

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Ring...Ring...

Ring...Ring...

Ring...Ring...

Ring...Ring...

Ring...Ring...

Ring...Ring...User1User1

User 2User 2

User 3User 3

User 4User 4

To coverageTo coverage


Describe how the proposed solution supports circular hunt groups.

The VCX solution supports single-site circular hunt groups in the current release. A circular hunt group is a cyclic pass through a list of members with timeout, and call coverage to a hunt group mailbox.

The routing of a circular hunt group call happens in round-robin fashion in a circular hunt group. The call will ring the non-busy phone (ringing phone is considered busy) to receive the call. Circular Hunt Groups save the ID of the member to whom the last call was routed to (doesn’t matter if that member picked up the call or not). The next incoming call therefore starts to call on the next member’s phone (if logged in) from the member list.

A call gets passed on to the next available member after a per device timeout. A logged out member is not called. When the routing reaches the last circular hunt group member, it again starts from the first hunt group member and routes down the member list until it is either answered by a member or total timeout elapses. In the event of total timeout the call is forwarded to call coverage path.

If all the members are busy then the circular hunt group waits until total timeout before forwarding the call to call coverage. In the meantime, if a member becomes available then the call is routed to that member.

If all members are logged out, the call waits till total timeout before being forwarded to call coverage. In the meantime, if a member logs in then the call is routed to that member. If there are no members in the circular hunt group, the call is forwarded to the call coverage path without waiting.


Ring...Ring...

Ring...Ring...

Ring...Ring...

Ring...Ring...

Ring...Ring...

Circular Hunt Group

User 4

User 3

User1

Ring...Ring...User 2



Describe how the proposed solution supports linear hunt groups.

The VCX solution supports linear hunt groups in the current release. A linear hunt group is a single pass through a list of members, with call coverage to a hunt group mailbox.

The routing of a linear hunt group call starts from the first member of the hunt group down to the last one for all the calls that come into the linear hunt group. If a member is on call/hold they will not get the call. Call ringing is also considered as busy.

A call gets passed on to the next member after a per device timeout. The call is forwarded to the call coverage extension either after calling the last member of the group or total timeout – which ever happens first.

The call is forwarded to call coverage path immediately when all the members are busy and also if there are no members in the Linear Hunt Group. A logged out member is not called. There is no call queuing for a linear hunt group.

Describe how the proposed solution supports last number redial.

The 3Com VCX IP Telephony solution supports the Last Number Redial feature. The Last Number Redial feature stores the last number dialed by the phone set user and allows the user to automatically dial the number by pressing a button on the phone set.

Each 3Com IP phone set has a “Redial” button. When the “Redial” button is pressed, the last number that was dialed by the phone set (for this user) will be automatically dialed. The Last Number Redial feature is also available by using feature code 401.

Describe how the proposed solution supports malicious call trace.

The VCX solution supports the Malicious Call Trace feature. Malicious call trace is a way for a user to alert the system that they have received a call that they feel is harassing. The VCX system is capable of:


Ring...Ring...

Ring...Ring...

Ring...Ring...

Linear Hunt Group

User1

User 2

User 3



1. Sending a trap to network management that indicates the malicious call. The trap will include the number of the user invoking the feature, the malicious calling party number, and the date and time of the event.

2. Log the Malicious Call Trace (MCT) feature in the CDR for the call. The system will add the MCT feature type to the “featureInvoked” portion of the CDR.

3. Use the gateway to send a facility message to the PSTN and alert the relevant authorities. The facility message will be sent by the Call Processor as a NOTIFY message with the following header included: X-ISDNTunnelingInfo: 621c0991a106020101020103

The Malicious Call Trace feature is invoked by pressing an administrator and user-mappable button, or using feature code 119. After invoking the MCT feature, the LCD of the phone that invoked the MCT will display “malicious call” for the duration of the call.

The typical use case is that the user on the system has received a call via the PSTN. In this case, the VCX system will provide all three of the MCT aspects noted above. Note that the gateway(s) will require additional configuration to provide this feature and not all gateways may support it.

The feature may also be used by a user who has received a call from another user on the system. In this case, the system will only provide the first two MCT aspects listed above.

The system does not place any restrictions on who may invoke MCT. It could be the called or calling party (if both are system users).

The system will not prevent conference call users from invoking MCT. Once MCT is invoked, it cannot be revoked or unmarked. There are no limits in the VCX system on how many simultaneous MCT features can

be invoked.

Describe how the proposed solution supports audio indications for Message Waiting Indicator (MWI).

The 3Com VCX IP Telephony solution provides a “stutter-tone” to indicate the user has a message waiting.

Describe how the proposed solution supports visual indications for Message Waiting Indicator (MWI).

310x 3Com IP phone sets provide a Message Waiting Indicator (MWI) light. The MWI light remains red as long as there are unreviewed messages in your mailbox.

In addition, all 3Com IP phone sets with displays support a MWI display that includes the number of new messages and total messages.

Describe how the proposed solution supports missed call indicator with callback.

The VCX IP Telephony module supports a Missed Call Indicator.

The 3Com IP phone sets have a missed call indicator on the display panel which indicates the number of missed calls. When a user returns to their desk to find a Missed Call Indicator



message, they can press a Soft button (Exit, right-most) to see a display of each missed call (starting from the most recent). For each missed call, the display will present the calling party information (if available), the date, and the time of the call.

A user can automatically dial the originator of a missed call by pressing a Soft button (Select, left-most) while viewing the missed call list. A user can also scroll through the list by using the Scroll buttons on the 3Com IP phone sets with displays.

Describe how the proposed solution supports the ability to use your phone settings on another phone.

The VCX IP Telephony module supports the Hoteling feature. Mobility is an inherent characteristic of the Session Initiation Protocol (SIP) and an integral part of the VCX IP Telephony solution. The Hoteling feature provides the ability for a user to login from any IP terminal anywhere in the network that is connected their home call processors and get the same feature set as their primary phone set.

310x users on the system have a username and password. The Hoteling feature allows users to login to any phone connected locally or remotely to the system. By logging out on a 3Com IP phone set connected to the system, a user can then log in with their username and password.

If the Multiple Contacts feature is enabled (Number of Contacts greater than one) on the VCX, the primary phone does not have to be logged out for a user to remotely login.

Logging out is performed by pressing “Program” + 5 + 6, or without a “Program” button using a feature code (*600 + 5 + 6).

From the phone that you want to use as yours, enter your phone number and password:

Press “Program” + 5 + 4 (or *600 + 5 + 4), enter your phone number, and then press #.

Press “Program” + 5 + 5 (or *600 + 5 + 5), enter your password, and then press #. When you are finished using the other phone, log out of the phone.

Describe how the proposed solution supports multiple music on hold sources with custom recordings. Are multiple music sources supported for differing groups or departments?

The 3Com VCX solution provides a scalable, efficient, and flexible Music on Hold feature. Music on Hold (MOH) allows callers to hear a particular recording continuously while on hold. The VCX Music On Hold feature allows administrators to assign specific MOH files to different groups of users on a per-phone basis. The VCX solution supports an unlimited number of MOH sources.

The MOH sources are stored at the 3Com IP Messaging module. When a caller is to be placed on hold, the VCX redirects the call to the IP Messaging module at a particular extension. The extension given to the IP Messaging module identifies the MOH source to play.



The MOH sources are downloaded to IP Messaging in .wav format using sftp or the 3Com Enterprise Management Suite. Once the wav file is downloaded to an IP Messaging server, it is converted to both G.711 and G.729 formats required for play by IP Messaging. The IP Messaging Operations and Administration Guide and the EMS VCX User Guide details the procedures for creating and downloading MOH source files.The directory on the IP Messaging server where the MOH files are located is /usr/app/app.dir/speak.vox, and the MOH files must be named with lower case and fewer than 8 characters.

Music On Hold is implemented using IP unicast and does not require additional bandwidth. In addition, the Music On Hold feature is implemented in the VCX call processor instead of the phone, providing improved functionality and feature interactions:

There are no CDR’s generated for Music On Hold call legs. Music On Hold functionality is supported for all endpoints, including FXS ports

Describe how the proposed solution supports the ability to mute a call so that the remote party or parties are still connected but cannot hear the user who initiates the mute?

The VCX IP Telephony solution supports the Mute feature. The Mute feature allows phone set users to prevent callers from hearing them while on a call.

The mute feature can be enabled by pressing a “Mute” button (which is a hard key on the phone) or by entering feature code 101. To disable the mute feature (allow caller to hear you again), press the “Mute” button or the feature code again.

The mute feature functions when the phone set user is using the hands-free (speakerphone) or if the receiver is off the phone set.

Describe how the proposed solution support night service (time of day) call routing.

Describe how your system supports the re-use of the same extension number in different offices where call processing servers are located.

Describe how the proposed solution supports paging from a phone to a loudspeaker paging system (integrate-able with external PA system).

Normally, the 3Com VCX solution interfaces to a paging system via an analog FXS media gateway. This allows users to dial an extension and then page overhead.

Describe how the proposed solution supports paging phone to phone.

The ability to page phone to phone is supported by the VCX IP Telephony module. This feature is also known as “send beep with calling name”. The VCX allows a user to beep another user via feature code 331. The called party will see the calling name on their LCD



along with a beep. This VCX beep feature is a page sent between phones, the called party’s phone will tell who beeped them.Describe how the proposed solution supports paging groups to IP phones.

The VCX IP Telephony module supports Paging Group functionality through the phones.

With the Paging Group feature, a caller can broadcast a message to other phones that are members of the same paging group. There can be up to 50 members per group and up to 800 groups per site (call processor).

To implement this feature, an extension is assigned to the paging group. When this extension is dialed, the speakers on all “available” phones in the group are activated and begin broadcasting audio from the caller.

Available is defined as:

Logged in Not in any existing calls (regardless of the state of those calls, for example: ringing,

hold, connected, etc) No redirect features active. Examples are: Call Forward, Do Not Disturb.

Invoker simply dials the number of the page group to begin the page. Audio heard via the page will be one way only. Recipients of the page will be able to disconnect the page either by using the speaker button to disconnect the page, or by picking up and replacing the receiver. The LCD display of both the sender and receivers of the page will display the number & name of the page group during the page.

To configure the Paging Group feature, an administrator creates a new group of type Page Group. The Page Group has the following properties:

Name Number Member List Multicast address Multicast port (optional and used only if populated. Otherwise, the paging phone will

pick a port)

The initial design utilizes multicast audio in order to provide the packet replication of the source audio. In this configuration, multi-site paging groups are possible, but the network



between sites will be required to be set-up such that the multicasts are able to traverse the network.

Additionally: Only one page is allowed within the group at one time If phones transition from an unavailable state to an available state while there is a

page ongoing, the phone will begin to hear the page Access to page feature and permissions for paging other users will be controlled by

the page group configuration, there will be no additional TOS or COS parameters related to paging

Examples of the provisioning screens for a Page Group are illustrated below.

Feature Interactions and Use Cases

Pages will not be bridged. If the primary user of a bridge uses a BSA line to send a page, the secondaries will see the bridge-line as busy, however the primary user will not be able to place the page on hold so there will be no shared hold capability.

Receivers of a page should not cause a BLF to illuminate, senders of a page should have BLF illuminated.

Pages cannot be forwarded, if forward universal is active, the phone will not be paged

If DND is active the phone will not be paged, if DND is activated while a phone is being paged, the phone should be disconnected from the page.

Page calls cannot be parked. Page calls cannot be picked up (directed pickup or group pickup). You cannot camp-on a page number Sender and receiver of a page cannot hold a page Sender and receiver of a page cannot transfer the page. A receiver of a page will be disconnected from the page when they hit the release

button. A sender of a page will stop paging to receiving phones when they hit the release

button. A member of the page group with HandsFree enabled will hear pages. A member of the page group with FwdMail enabled will NOT hear pages.



Sender and receiver of a page can NOT conference a page.

Users can view the Paging Groups they are members of by using the VCX VoIP User web provisioning interface, as illustrated in the example below.

Describe how the proposed solution supports the ability to silently monitor and barge-in to an established connection.

The 3Com VCX solution supports the Silent Monitor/Barge-in feature, which allows a user to enter into an established connection. After barging in, there is a display on the phone being barged in on that the barge-in is in process. There is no warning tone when a user barges in.

The Silent Monitor feature is invoked by using pressing an administrator-mappable button, or using feature code 425. A user can only perform silent monitor/barge-in when configured by a VCX administrator through an Access Control List via web provisioning. The Barge-in feature is invoked by pressing an administrator-mappable button, or using feature code 428.

The VCX silent monitor/barge-in feature functions in a multi-site environment. A typical use case would be: Supervisor on site A monitors a call connect to an agent on site B.

Describe how the proposed solution supports speed dialing.

The 3Com VCX IP Telephony supports the Personal Speed Dialing feature, which enables one touch dialing from mappable buttons or feature code.



Each VCX user can have up to 12 different speed dial numbers for each extension. Each speed dial number can contain up to 32 digits. Personal Speed Dials are invoked by pressing an administrator and user-mappable button, or feature code 601 plus the speed dial number. Personal Speed Dials can be configured on the phones using feature code 601 by entering the speed dial number followed by the number to be dialed.

Personal speed dial numbers can also be configured using the VCX VoIP User web provisioning interface. The following screen shot is an illustration of the VCX VoIP User web provisioning interface for the Speed Dialing feature.

Describe how the proposed solution supports Busy Line Field functionality.

Speed dial buttons on 3Com IP Phones have Busy Line Field functionality, displaying the status of the extension mapped on the device.

This feature can only be configured by a system administrator. A red dot next to a Speed Dial button indicates an administrator has configured the status light for this button so that it indicates when the target extension of the speed dial is in use. The status light will blink when the target extension is in use.

Describe how the proposed solution supports transfer directly to voice mail for any mailbox.

The VCX IP Telephony module supports the Direct Transfer to Voice Mail feature.

This feature allows users to transfer a caller directly to the mailbox of any other user on the system. The Direct Transfer to Voice Mail feature works within a single site or across multiple sites. The Direct Transfer to Voice Mail feature is invoked by pressing an administrator and user-mappable button, or by using feature code 441.

Can a phone be configured to dial a preset number after the phone has been offhook for a set period of time (warm ring down)?



The 3Com VCX IP Telephony module supports the Warmline feature. The Warmline feature allows a 3Com IP phone to connect to a pre-determined number if the user picks up the phone and does not dial a number within a pre-determined time period.

The preset number and the offhook time is configured by administrators, and is automatically dialed when the phone is offhook for the configured period of time.



2.10 Unified Messaging

Briefly describe an overview of the messaging solution included with the proposed solution.

As an integral component of the 3Com Convergence Applications Suite, the 3Com IP Messaging module helps to reduce costs by replacing proprietary voice mail equipment with a network server-based solution and centralizing administration of the application. By providing centralized applications and key features that enhance contact with employees and customers, the 3Com IP Messaging module also helps to increase user productivity.

The 3Com IP Messaging module has several key differentiators for enterprises looking for a robust, centralized IP messaging system; including:

All-IP SIP based solution Full-featured voice messaging and unified communications Advantages of centralized administration and global user directory Inter-operability with legacy PBX’s High availability with geographically dispersed redundancy Scalability at all levels

Using the Session Initiation Protocol (SIP), the 3Com IP Messaging module integrates seamlessly with the VCX IP Telephony module and easily scales from one to thousands of mailboxes. Reliability is a key part of the architecture, with several supported redundant configurations to minimize and avoid service outages. The 3Com IP Messaging module provides auto-attendant functionality, incoming fax mail, and find me follow me features. The 3Com IP Messaging module provides email integration using SMTP, POP3, or IMAP4 in auto-delivery or synchronized unified messaging configurations.


SMTP, POP3, IMAP4

DominoEmail Server

IP Messaging

Lotus Notes


When a mailbox owner accesses their mailbox via an e-mail client, their voice, fax, and e-mail messages are available. Voice messages are .wav attachments, and fax messages are .tif attachments.

The 3Com IP Messaging module supports the Email Synchronization feature, which represents true unified messaging for subscribers. Their email client interfaces only to the email message store, and they have synchronization of all or voice mail only messages and message waiting indication at the voice mail store. This is accomplished using a combination of auto delivery and periodic polling with the email message store.

When new voice or fax messages are received, they are auto delivered to the email message store by IP Messaging via SMTP. When messages at the email client are read or deleted, the associated user’s phone will have their message waiting indication adjusted accordingly. You can choose to synchronize all messages (including emails), or just synchronize voice and fax messages. Since IMAP4 provides the status of unreviewed/reviewed messages, and POP3 does not, IMAP4 provides the fullest level of email synchronization. The poll interval can be configured for each mailbox and the IP Messaging module automatically polls at mailbox login time. When the email password changes, the IP Messaging module detects this and sends a System Message to the mailbox owner informing them to change their email password in their mailbox, and disables polling until the password is changed and validated.


Email Client

SMTP

MWI

SMTP, POP3, IMAP4

Voice & Fax Messagesin your email client

POP3, IMAP4POP3, IMAP4

MWI synchronizedafter each poll interval

OptionalText To Speech

Server

Email Text

DominoEmail Server


Describe how the proposed solution can support a distributed unified messaging solution.

Global Voice Mail is an IP Messaging feature that allows a group of individual IP Messaging systems to act as a single unit from the user’s perspective.

When do you use Global Voice Mail?

When a distributed voice mail solution is required o Due to multiple locations, IP network, scalability, and resiliency requirements

When centralized provisioning and a global scope for users is required Provide a truly enterprise-wide messaging solution

What do you get with Global Voice Mail?

Global Name Directory Global Name Announcements Global Message Sending Global Provisioning

The 3Com IP Messaging module (IPM) provides an enterprise-wide Global Messaging solution within the cost, scalability, and resiliency requirements of each location in the enterprise. Individual IP Messaging systems can be deployed in a distributed fashion across the enterprise, each one serving a sub-set of the total user base. The messaging data for each user is maintained on the “home” IPM system, which can be any type of IPM system. The directory data for all global users is shared and synchronized across each individual IPM system, allowing users to access anyone in the global user directory regardless of their home location.

IPM system types include standalone, Dual All-In-One, Dual IPM, and client/server, with each type having its own redundancy and scalability characteristics. By using a Central Server to coordinate provisioning of user profile data to each individual IPM system, the 3Com IP Messaging module can play the name announcement of any user on the system and forward messages to any user in the system - regardless of their home IPM system.

The IPM Global Messaging functionality operates efficiently in the background in near-real-time as IPM data tables are modified and updated in either direction. The central server is implemented as a Dual IPM system providing only this functionality.

IPM Global Messaging supports centralized provisioning of user mailboxes, locations, and class of service. Provisioning can be done at the central server and is automatically synchronized with each local office. In addition, provisioning can be done at a local office and is automatically synchronized with the central server, and then to the other local offices as needed. Local offices synchronize the global directory table, but only synchronize data that is needed locally or requested.

The IP Messaging Global Voice Mail feature is based on the concept of the IPM Global Directory Table, which is the VCX Global User Directory extended to IP Messaging. The IPM Global Directory Table is synchronized to each IPM Local Office via an IPM Central Server.



The IP Messaging Global Voice Mail feature is primarily based on the global directory table (vgdd_dir) which is replicated across all the local offices. The table contains several different record types i.e. Site, Mailbox, COS, Company, and SUG. Each Local Office periodically polls the table on the Central Server by doing an External Server lookup on the modifytimestamp field. Whenever a record is modified, the modifytimestamp is updated so that all the local offices will lookup the changed record on their next poll. When a record changes, the local offices check the type and process accordingly, then write the record to their local copy of the table.

The Mailbox type record contains all the info needed for users to send messages to mailboxes on remote offices including NA filename and timestamp, Firstname, Lastname, and the dtmf lookup fields for dialbyname. Subscriber profiles are stored on the local office they belong to and the Central Server. The only way a local office knows about mailboxes on other remoted offices is through the global directory table.

The 3Com IP Messaging Global Voice Mail feature supports these features:

Global Name Directoryo Allows subscriber to lookup by name any other Global Directory enabled

subscriber on any IPM Local Office associated with the same Central Server Global Name Announcements

o Subscriber name announcement recordings are available globally across all IPM Local Offices

Global Message Sendingo Allows a subscriber to send messages to subscribers on other IPM Local

Offices Global Provisioning

o Mailboxes can be provisioned on the Central Server and automatically propagated to the appropriate IPM Local Office

o Provisioning can also be performed at the IPM Local Office with the changes being automatically propagated up to the Central Server


Local Office(Regional)

IMIM

Local Office(Branch)

IMIM

Central Server


IMIM


IMIM


IMIM

Local Office(Branch)


Outline if any of the license charges are a recurring expense, and in the case of the Voice Mail Seat License how additional or less voicemail users on the system are handled administratively through billing and maintenance costs.

The 3Com licenses stated in the proposal are one-time, non-recurring charges. This does not include service and maintenance costs.

For VM seat licenses, the actual number of seat licenses for the initial proposal can be adjusted to be more or less based on discussions with the University. The service and maintenance costs will be reflected in the proposal for the desired number of seats. For additional licenses, a quote would be provided by 3Com for the desired number of seats and for the maintenance costs associated with the seats. There is generally no service cost associated with adding VM seat licenses, unless specific work is requested from 3Com.

Describe how the proposed solution supports the ability to “read” email messages to users.

As an optional purchase, the Text To Speech (TTS) feature allows 3Com IP Messaging module applications to speak words based on text strings. The 3Com IP Messaging module integrates third party TTS engines to perform this functionality. The TTS feature is available for G.711 and the English and Spanish (Latin America) languages, and can be deployed based on network requirements and number of ports.


ScansoftSpeechWorks

Speechify

TTS Server

ScansoftSpeechWorks

Speechify

Voice Engine

Text

Speech

3ComIP Messaging

TTS Adapter

Text

Speech

Windows XP or Windows Server

3ComIP Messaging

Call Builder

IP Messaging ServerAuto-delivery of

voice/fax messages(SMTP)

Email message send, reply, forward

(SMTP)

Email message or MWI synchronization

(POP3/IMAP4)

TextSpeech

SIP

Email Client Email Server


Explain how fax calls are supported with the proposed unified messaging solution.

The 3Com IP Messaging module provides strong inbound fax capabilities, with the ability to receive fax calls into any user’s mailbox and auto-deliver the fax message to an email account, network printer, or a fax machine. Incoming fax calls can also be directed to a fax machine connected to an analog FXS media gateway.

The 3Com IP Messaging module provides a Never Busy Facsimile feature by using the Facsimile Message Deposit feature along with the Facsimile Auto Print feature to record new inbound facsimiles, and automatically printing each one in the order/priority they were received.

The following diagram illustrates how an incoming fax is routed to IP Messaging, and shows the different ways that IP Messaging can deliver the fax message.

Describe the capabilities of the proposed unified messaging solution to support multiple languages.

The 3Com IP Messaging module contains support for multiple languages and dialects. Voice prompts and messages are recorded in the appropriate language, and application scripts access these messages by setting a system variable to the appropriate language.

Application scripts sometimes ask end users which language they wish to use, while other application scripts use default language information stored in a subscriber’s profile. Speak files for each language are stored under their own directories.

An unlimited number of languages can be simultaneously supported by the 3Com IP Messaging module, and in fact, many more are planned for subsequent releases. In the current release, the following languages are supported by the 3Com IP Messaging module as standard offerings:

Language Available in G.711? Available in G.729?English (American) √ √English (U.K.) √ √Spanish (LAT) √ √French (France) √ √


Analog FXSGatewayDigital

Gateway

IP NetworkIP Network

VCXIP Telephony

Fax Machine

NetworkPrinterIP

Messaging

EmailClient

SMTP,POP3, IMAP4

DominoEmail Server


French (Canadian) √ √Italian √ √Chinese (Mandarin) √ √Portugese (Brazil) √ √

With separately purchased custom integration, other languages and dialects can be supported by the 3Com IP Messaging module. The recordings for the voice prompts must be performed by either a 3Com voice talent or the customer, the TUI must be modified to support the grammar and syntax of the language, and a new identifier for the language must be added to the 3Com IP Messaging module.

Can subscriber mailbox features be reset without loss of voice messages?

System administrators with mailbox permissions have the ability to reset mailbox features without the loss of voice messages.

Can a subscriber mailbox be reinitialized without loss of voice messages and custom greetings?

System administrators with mailbox permissions have the ability to reinitialize a mailbox without the loss of voice messages and custom greetings.

Can subscriber mailboxes be reinitialized with all messages and greetings deleted?

System administrators with mailbox permissions have the ability to reinitialize a mailbox and delete all voice messages and custom greetings.

Is there a programmable feature to disable a mailbox after a number of unauthorized attempts have been made?

The 3Com IP Messaging module allows an administrator to configure a mailbox to be disabled after some configurable number of unauthorized attempts have been made.

What options are available if a mailbox receives more than the number of messages allowed?

When a mailbox is full, the caller will hear the following recording, and then the 3Com IP Messaging module will initiate disconnect of the call:

“This mailbox is temporarily unavailable.”

This is configurable by administrators to play no warning, or a warning when the mailbox is full in 5% increments from 55% to 95%. To override, an administrator would change this configurable to no warning.Does the system perform automatic housekeeping routines which free up disk space by purging messages after a pre-defined period of time?

The 3Com IP Messaging module performs automatic housekeeping functions by automatically purging mailbox messages after administrator-defined intervals.



What is the maximum number of access ports that can be supported on the proposed solution?

The concept of “ports” does not apply to an IP messaging service since the interface is not based on TDM technology. Since the 3Com IP Messaging module is IP-based using SIP, the number of ports is a factor of number of processors, memory, and IP network design.

In PBX integration configurations, physical ports that essentially limit the number of call sessions are defined by the number of T1 spans or digital/analog lines that are configured on VCX media gateways which connect to PBX’s or switches.

The number of ports that can be active at any one time varies with the server type, server configuration, number of processors, etc, and ranges from 1 to 200 active simultaneous call sessions.

The VCX system is sized based on server capacity, system architecture and seat licenses. The highest capacity is defined in a client-server architecture in which the MMU and MSU are resident on physically separate servers. The client/server architecture supports 20,000 – 200,000+ users with up to 2,000 port capacity for simultaneous calls.

Describe how messages can be automatically purged.

Purging of messages from mailboxes is automatic and supports the following configurable parameters:

Expiration for new messages (days) Expiration for saved messages (days) Expiration for deleted messages (days)

Is there the capability to send a system message without sending message notification?

The 3Com IP Messaging module allows administrators to send system messages with or without message notification.

Can the system administrator limit the ability of users to request outcall notification?

Administrators can limit the ability of users to request outcall notification.

What is the maximum number of times the system will attempt an outcall to a predefined destination number?

Up to 999. Configurable via VMAdmin and web provisioning.

Are mailbox pass codes concealed from the system administrators?

The 3Com IP Messaging module does not reveal the value of mailbox passwords in the system administration interface used by administrators.



Provide and overview of the reporting and logging capabilities of the proposed unified messaging solution.

The 3Com IP Messaging module provides a robust set of system logs that are available to system administrators. These system logs include app.out, listen.out, Call Detail Records, and vmlog.dir files.

A sample of the vmlog.dir log files is shown below:

rec_type=”vlep_log”,vl_key=”Jan 19 2005 11:16:35.00”,vl_sec=1106154995”,vl_who=”30001:VMCA_PG “,vl_object=”8472628534”,vl_what=”ani=’8478432000’ entering personal greeting”

SMTP client process logs to /usr/app/gen/eml_client.out.

Describe system reporting capabilities.

The 3Com IP Messaging module provides a robust set of reports that are available to system administrators. Detailed descriptions of these reports, including samples, are provided in the attached VCX Technical Information document titled “3Com IP Messaging Reports”.

The 3Com IP Messaging module provides reports on a demand basis or on a scheduled basis. All reports can be viewed in real time on the XTerminal VMAdmin GUI or written to a disk file.

Many reports can be delivered to one e-mail account. Reports can be scheduled on the following basis:

Daily at a particular time Once a week on a particular day at a particular time 1st of the month at a particular time 15th of the month at a particular time 1st and 15th of the month at a particular time

The following reports can be generated by the 3Com IP Messaging module:

Report Name Report can be Scheduled?

Report to Screen?

Report to File?

Report to E-Mail?

All Subscribers No Yes Yes NoSubscribers by NPA No Yes Yes NoSubscribers by NPA-NXX No Yes Yes NoSubscribers by Company/Division No Yes Yes NoAll Classes Of Service No Yes Yes NoID Class Of Service No Yes Yes NoAdministrators No Yes Yes NoMessage Time Statistics No Yes Yes NoPort Usage Statistics Yes Yes Yes Yes



Report Name Report can be Scheduled?

Report to Screen?

Report to File?

Report to E-Mail?

Mailbox Usage Statistics – Regular Mailbox

Yes Yes Yes Yes

Mailbox Usage Statistics – Auto Attendant

Yes Yes Yes Yes

Disk Usage by Message Type Yes Yes Yes YesIdle Mailboxes Yes Yes Yes YesActive Mailboxes Yes Yes Yes YesCall Processing by Mailbox Yes Yes Yes YesCall Processing by Port Yes Yes Yes YesInitialized Mailboxes Yes Yes Yes YesFailed Login Attempts Yes Yes Yes Yes

Briefly describe the auto attendant features and functionality of the proposed IP messaging system.

Auto-attendants provide the ability to perform menu prompts and touch-tone interaction with users. They can easily be customized by <<clientShort>> administrators to provide any number of user interaction and menu prompt sessions.

There is no hard limit to the number of auto attendants that can be created and used. Each auto-attendant supports up to 99 individual nodes. The 3Com IP Messaging module allows up to 10 transfers per node (leaving *, #, and timeout for other functions). With a multi-node auto attendant design, this can translate into literally hundreds of transfers allowed per auto attendant.

The 3Com IP Messaging module can use its auto attendant functionality to provide the ability for administrators to "post" messages and for callers to hear them. This allows only those with password permissions to the auto attendant to post messages.

To create an auto attendant, administrators use the Speak utility to record their own prompts, and use the VMAdmin auto attendant screens to define the characteristics of the auto attendant. This includes the definition of which DTMF entries are valid and their function (go to a different auto-attendant node, another auto-attendant, transfer a call, speak a recording, etc.). The auto-attendants support a time schedule capability and the ability to execute a customized script.

Are different automated attendant greetings available on a per application basis by: Time of day, Day of week, Weekend and holiday, and Exception days?

The 3Com IP Messaging module allows multiple auto attendants to be configured on the same system with different automatic schedules for TOD, DOY, and holiday schedules. Each auto attendant can have its own automatic schedule.

Can multiple auto attendant applications run concurrently on the same system?

The 3Com IP Messaging module supports the ability to configure and concurrently run multiple auto attendant applications. There is no hard limit to the number of auto attendants



that can be configured and running at the same time, but is limited by number of access ports and server configuration.

Can integrated voice mail, fax, auto attendant run on the same interface port?

All ports that connect to the 3Com IP Messaging module can be used for either voice mail, fax mail, or auto attendant. The ports do not have to be assigned for any particular use, as all ports can be used for voice, fax, and auto attendants.

Are directory options available on a per application basis?

The 3Com IP Messaging module provides a directory search for several different functions, including:

Message Forward Message Send Auto Attendant Directory Assistance

The 3Com IP Messaging module supports a Dial By Name directory search function. For the directory search by name, a caller enters touch tone keys representing the mailbox owner’s last name or mailbox number. The 3Com IP Messaging module searches for a match to the input and presents the results to the caller. The caller selects the appropriate result by pressing the # key.

This feature speaks the mailbox names to the caller using the recorded name announcements or text-to-speech if enabled. If a name announcement is not recorded and TTS is not enabled, the mailbox number will be spoken to the caller.

If several directory entries match the user input, then all matching entries are presented to the user with a numeral for each. The user then listens to the matching entries and makes their selection by entering the appropriate numeral.

The 3Com IP Messaging module supports the ability to have different directory options available on a per application basis or on a system-wide basis.

This is implemented using Send User Groups and by defining individual mailboxes to certain auto attendants.

Are all access ports available to all unified messaging interfaces, i.e. outcall to pager, pda, cell, etc?

All ports are available for auto attendants, all types of mailboxes, unified messaging interfaces, outcalls, and incoming calls.

Can either ports or storage be added to the system, without requiring that the Voice Mail system be taken out of service?

The proposed 3Com solution includes enough capacity to support the initial configuration plus growth. Since the proposed solution is a fully redundant solution, it is possible to



maintain service in a stand-alone mode while servers or storage capacity are added. The increased capacity is added at one redundant node at a time; then when complete, the system is restored to full redundant mode.

Briefly describe personal distribution lists.

Mailbox owners can create up to an administrator-defined number (up to 9,999) of distribution lists for their own private use via the TUI or Web Provisioning. For each entry in a distribution list, the subscriber can define a mailbox number, external phone number, and e-mail address for the recipient. The maximum number of entries in each distribution list is configurable by an administrator (up to 999,999).

Each personal distribution list is identified by a 1 to 4 digit number. Entries in a personal distribution list may be internal or external phone numbers.

Can non-system members be a part of a distribution list?

The 3Com IP Messaging module supports distribution lists that can have the following types of entries in the list:

Mailbox on the system External phone number (including non-system members) Email address

Does the proposed IP messaging system support the ability to broadcast a message to a group of users?

Broadcast lists are provisioned by an administrator with system privileges as well, but should be private lists available only to the system privileged mailbox they are created in. Broadcast lists can use COS or Company or All Subscribers as destinations and can be used to send System messages to a large group of subscribers from a system privileged administrator mailbox.

Broadcast lists are used by the Message Broadcast feature which allows a mailbox owner with system privileges to send a message to many destinations, which can include internal phone numbers, external phone numbers, class of service, company, all subscribers, personal distribution lists, and system distribution lists.

This feature is typically used by system administrators with system privileges to send System Messages to a large group of users from a system privileged administrator mailbox. The broadcast list is private to the mailbox they are created in.

Can the system generate a fax cover sheet for the receiving user?

The 3Com IP Messaging module does not automatically generate a fax cover sheet for incoming calls. When sending or forwarding fax messages, the 3Com IP Messaging module only inserts a cover sheet when defined by the user.



Are all features that apply to voice messages (private, future delivery, etc.) available for fax messaging?

The same treatment options that are available for voice messages are also available for fax messages.

Can callers and users attach a voice message to a fax?

The 3Com IP Messaging module supports the ability for users to attach a voice message to a fax being sent from a mailbox.

Can users choose to print a fax to any fax machine interactively?

Users have the ability using the TUI to print a fax to their pre-configured fax machine or to any fax machine interactively.

Can users choose to print all unprinted fax documents at once?

If the mailbox owner is calling from a fax machine handset, they can press start while they are reviewing a facsimile message in their mailbox, and the 3Com IP Messaging module will subsequently print all unreviewed facsimile messages currently in their mailbox using the delivery options configured for their mailbox.

Does the system support fax broadcast capabilities?

Users can send a fax message to a distribution list using the 3Com IP Messaging module.

Does the system support fax-on-demand capabilities?

As part of the Message On Demand feature, the 3Com IP Messaging module supports fax on demand capabilities.

Can the system automatically delete a fax after it is successfully printed?

A mailbox can be configured to automatically delete a fax after it is successfully forwarded or printed. Can users choose to receive a fax from the fax machine from which they are calling?

If the mailbox owner is calling from a fax machine handset, they can press start while they are reviewing a facsimile message in their mailbox, and the 3Com IP Messaging module will subsequently print all unreviewed facsimile messages currently in their mailbox using the delivery options configured for their mailbox.

Can a mailbox be configured with several "back-up" telephone numbers that are automatically dialed if the primary number is not answered?



The FindMeFollowMe feature gives mailbox owners the ability to make contact with others more efficiently regardless of location, day, time, or network access device. The FindMeFollowMe feature works across many types of networks including PSTN, wireless, and VoIP. The FindMeFollowMe feature supports simplified and complex contact methods.

The primary component of the FindMeFollowMe feature is the mailbox owner profile. Using the web, Telephone User Interface (TUI) mailbox owners create a list of locations where they can be reached, a list of favorite contacts, and a list of rules with different priorities, filters, and an operation. The rules define how others can find and reach the mailbox owner for each of several pre-defined events. There is no hard limit to the quantity of numbers that can be configured for the system to search through.

Does the proposed IP messaging system support forms (Q & A) mailboxes?

The 3Com IP Messaging module provides forms or Q&A functionality through the Forms mailbox feature. This feature allows administrators to setup the mailbox prompts and specify whether input is voice recording or DTMF response. Administrators and subscribers with access to the Forms mailbox password can then login to the mailbox to collect the data.

Describe Informational (Listen Only) mailbox functionality.

The 3Com IP Messaging module provides Listen Only (or Information) functionality through the Message On Demand feature. Message on Demand refers to a tree of auto-attendant nodes that allow callers to navigate through the tree selecting to print faxes or listen to recorded voice files.

The Message on Demand feature is available on all mailboxes, and all subscribers have access to the feature if configured.

How many greetings does the mailbox support at one time? Can users edit/change personal greetings at any time?

Each mailbox on the 3Com IP Messaging module can have the following personal greetings:

One normal greeting One extended absence greeting One busy greeting Up to 9 personal greetings can be recorded for scheduled greetings

Describe the default system greeting used when a greeting has not been recorded by a mailbox owner. Can users choose or be required to use a standard system greeting instead of a personalized one?


office cell home

Voice MailVoice Mail

Incoming call


If no greetings have been recorded by a mailbox owner, the standard 3Com IP Messaging module greeting that is played when calls are forwarded for message deposit is:

“Please leave a message for mailbox number <mailbox digits spoken>.”

The 3Com IP Messaging module allows users to choose to use a standard system greeting instead of a personalized one by simply not recording a normal greeting if the tutorial was not used to initialize the mailbox settings or by deleting the normal greeting after it has been recorded.

Describe the personal scheduled greeting functionality.

Scheduled greetings are used to personalize how a mailbox is presented to callers leaving messages based on date and time characteristics defined by a mailbox owner. Mailbox owners can define scheduled greetings using the TUI or Web Provisioning, but greetings themselves can only be recorded via the TUI.

The 3Com IP Messaging module allows mailbox owners to define a scheduled greeting based on day of the week, time of day, all day, or a specific day. There is no limit to the number of schedules that can be created. Each schedule is assigned to one personal greeting that is recorded by the mailbox owner via TUI. Up to nine personal greetings can be recorded for any one mailbox (the nine include the extended absence greeting and busy greeting).

Personal greetings may be changed by mailbox owners when logged into their mailbox at any time by pressing 2 from the Mailbox Setup Menu.

Describe the personal busy greeting functionality.

The busy greeting is recorded by a subscriber when they login via the TUI and enter the greetings menu. The 3Com IP Messaging module will play the busy personal greeting (if recorded) when the call forward reason indicates “busy”, even if an extended absence greeting is active.

Does the proposed IP messaging system support the ability for mailbox owners to record and use an extended absence greeting?

The extended absence greeting is recorded and activated by a mailbox owner when they login via the TUI and enter the Setup Greetings menu. The 3Com IP Messaging module will play the extended absence greeting (if recorded and activated) when the call forward reason indicates “no reply”, “unconditional”, or “busy” (if no busy personal greeting is recorded). Callers can also leave a message when the extended absence greeting is played.

The extended absence greeting over-rides any personal schedule entries that match the current date/time. When the extended absence greeting is activated, the 3Com IP Messaging module will inform the mailbox owner that the extended absence greeting is activated whenever they login to their mailbox via the TUI. The extended absence greeting must be de-activated by the mailbox owner via the TUI.



Can a caller escape to their own mailbox when getting another person’s mailbox?

The 3Com IP Messaging module provides the ability for a caller to escape to their own mailbox when they are getting another user’s mailbox.

This is done by pressing the * key to back out of the other person’s mailbox, then pressing # to get the prompts for your mailbox and password. When mailbox number and password have successfully been entered, the caller is logged into their own mailbox.

Can callers exit the mailbox at any time to obtain assistance (during the greeting, after the greeting, before recording a message, or after recording a message)?

The 3Com IP Messaging solution allows callers to exit a mailbox at any time by pressing * to exit out of the current function. The * key can be pressed at any time such as during a greeting, after a greeting, before a recording, during a recording, and after a recording.

In addition, the 3Com IP Messaging module supports “zero-out” functionality that allows callers to be connected with an operator, an internal phone number, or an external phone number. Callers can press “0” at any time during menu prompts to be connected to a phone number that is configurable by administrators and mailbox owners.

Mailbox owners can change the phone number via TUI or web provisioning. Administrators can change the phone number via Admin or web provisioning.

Are callers and users notified that a mailbox is full? If so, how?

The 3Com IP Messaging module Mailbox Full Alert feature provides the ability to inform mailbox owners when their mailboxes become too “full”. This means that the storage allocation for their mailbox is about to be reached or has been reached.

The 3Com IP Messaging module will play a prompt informing the user they must delete New and Saved messages or they will be automatically deleted by the system. The 3Com IP Messaging module informs the mailbox owner of the number of days before messages will be automatically deleted.

New messages will always be allowed to be recorded and placed into the New Messages folder, but messages cannot be placed into the Saved folder until there is enough storage in their mailbox.

Describe the methods and procedures that are used for logging into a mailbox.

The Mailbox Login feature is used to obtain, collect, and authenticate the identity of mailbox owners who are attempting to enter their mailbox. There are multiple methods that a mailbox owner can use to login to the 3Com IP Messaging module, depending on deployment configuration and user profile configuration.

All methods use the following call information:



Called Party Number Calling Party Number (ANI) Redirecting Number (if present)

The general philosophy of the VCX IP Messaging Mailbox Login feature is to provide a user with simple and fast access to their mailbox by automatically detecting the user’s identity whenever possible. If a user’s identity is not available automatically to the 3Com IP Messaging module, it will always prompt for mailbox number first, followed by password.

Can users login to their mailbox without entering a password?

The Auto Login feature is used to automatically enter a mailbox, bypassing password entry. The feature is used when it is enabled for a mailbox or Alias and any of the following conditions occur:

For calls that are re-directed to the 3Com IP Messaging module (i.e. Redirecting Number is present):

o Calling Party Number must equal the Redirecting Number, which are both the same as the mailbox number

o Calling Party Number is an alias associated with the mailbox that is specified in the Redirecting Number

For calls that are directed to the 3Com IP Messaging module (i.e. Redirecting Number is not present):

o Calling Party Number is a mailbox or an Alias associated with a valid mailbox

The Auto Login feature can be provisioned by administrators and mailbox owners using the web provisioning interface.

Can mailbox owners annotate a message before forwarding?

When forwarding a message, a mailbox owner can annotate the original message by recording a message that is sent with the original message during the forward. The forwarded destinations will hear the annotated message first, followed by the original message.

Can callers append to their messages? Can users add comments on an already recorded message without re-recording the entire message?

The 3Com IP Messaging module allows callers to append (add comments to an already recorded message) to their messages when sending a message within a mailbox. This function is not available in the call answering portion of the TUI (message reply).

Does the proposed IP messaging system support the ability to automatically deliver voice messages to another phone?

The Message Auto Delivery feature allows mailbox owners to have new messages automatically delivered to another internal or external phone number.



Does the proposed messaging solution support message Auto playback?

When there are multiple messages in a folder to be reviewed, the Message Auto Playback features plays them all in sequence without any interruptions or prompts between each message.

Describe the features available for mailbox owners to delete a message. Can the system warn users of impending message deletion because messages have reached the allowed retention time?

During message review, a mailbox owner can move a message into the Deleted Messages folder.

The messages will stay in the Deleted Messages folder up to the maximum number of days allowed by the administrator. After the maximum retention period, the 3Com IP Messaging module automatically purges the message from the Deleted Messages folder. A value of zero days indicates messages are not to be automatically purged at all.

The 3Com IP Messaging module does not support the ability for users to be warned of impending message deletion.

Does the proposed IP messaging system support the ability for mailbox owners to retrieve a previously deleted message?

Messages that are in the Deleted Messages folder can be reviewed and moved to the Saved Messages folder by a mailbox owner at any time up until the maximum retention period for messages in the Deleted Messages folder.

Does the system support a different deletion schedule for new and saved messages?

The 3Com IP Messaging provides independent retention parameters for new, saved and deleted messages. When the retention period expires for a new or saved message, the message is moved to the Deleted folder. When a deleted message expires it is removed from the system.

Does the proposed IP messaging system provide the ability for mailbox owners to receive a confirmation that a message was delivered?

The Message Delivery Reports feature allows confirmation that a recorded message has been accepted or received by the 3Com IP Messaging module, when it is sent to a recipient, when it is delivered, whether delivery failed or is still in progress. This feature confirms the message delivery with the time and the date.

The report can be made via voice, facsimile, or e-mail. The report is made using personal setup configuration or default system configuration made by administrator.



The 3Com IP Messaging module does not support the ability to determine if a message has been listened to if return receipt is not requested at the time the message was sent.

Can the proposed IP messaging system deliver messages to non-subscribers?

The Message Delivery to non-subscribers feature provides a method for subscribers to send voice and facsimile messages to non-subscribers. This feature makes an outdial to the subscriber-specified number, and when connected, plays the following system prompt:

“You have a message. Press # pound to listen, or hang up.”

If the non-subscriber enters #, the 3Com IP Messaging module will speak the message recorded by the subscriber. The non-subscriber has the ability to rewind and fast forward through the message, and to reply to the message. The message is deleted from the system after the call with the non-subscriber is completed.

Describe the options available to callers when a message is deposited into a mailbox.

After recording a voice message, a caller has several options available if they did not hang up. The following options are available after recording a voice message:

Press 1 to replay o Allows caller to hear the message they just recordedo Standard message playback control options are available

Press 2 to re-recordo Allows caller to re-record message

Press 3 to mark message as urgento Message flagged as urgento Used to put in front of message queue and for notification procedures

Press 4 to mark message as privateo Phone number of caller will not be available for mailbox owner to review

Press 5 to enter call back numbero Allows caller to enter a number that will be played back to the subscriber

when they review this message. o This option only presented to caller if mailbox owner has callback feature

turned on Press * to cancel

o Deposits message in mailbox, disconnects call Press # to finish

o Deposits message in mailbox, disconnects call

Does the system store messages in separate queues such as Urgent, Unplayed, Saved, and Return Receipts?

The 3Com IP Messaging module supports the standard folders of Inbox, Saved Items, Deleted Items, Sent Items, and Drafts. The 3Com IP Messaging module also supports user-defined folders when used with IMAP4 clients.



The TUI provides access only to Inbox, Saved Items, and Deleted Items folders. Messages sent through the TUI are placed in the Sent Items folder if it is enabled. If the Saved Items folder is disabled, then the New and Saved TUI queues are mapped to the Inbox.

Future Delivery messages are placed in the Drafts folder until they are sent.

Return Receipt (Delivery Report) messages are placed in the Inbox.

Describe the features available with the proposed messaging system to forward a message. Can users choose to remove prior introductory comments before forwarding the message again?

During message review, a mailbox owner can forward a message to one or more destinations. An unlimited number of destinations can be entered by the mailbox owner who is forwarding the message.

A destination can be an internal phone number, external phone number, personal distribution list, or system distribution list. A destination can also be chosen using a Directory Search.

As a message is forwarded from mailbox to mailbox, all recorded annotations and the original message are retained and propagated intact. The 3Com IP Messaging module does not support the ability for users to remove prior introductory comments before forwarding the message again.

Describe the message priority levels available with the proposed IP messaging system.

Messages in 3Com IP Messaging folders are sorted based on the following criteria:

Urgent/normal/private Date Time Reviewed/un-reviewed

Describe the options available while recording a message.

After recording a message, the VCX IP Messaging module provides the following options to users:

Review recordingo The message recording is played back to the caller

Re-recordo The caller is prompted to record the message (replaces previous recording)

Append recording Rewind playback Pause playback Forward playback



Describe the features available with the proposed IP messaging system to reply to a message. When replying to a message, can a copy of the reply be sent to a user or group of users?

During message review, a mailbox owner can reply to the sender of a message in one of two ways:

Replying with a live call Replying with a message

The 3Com IP Messaging module allows message replies to VCX users and non-VCX users. If the sender information is available to the 3Com IP Messaging module, the mailbox owner will be prompted to reply to that phone number. The mailbox owner can enter a separate reply phone number regardless of whether the sender information is available.

For a reply with a live call, the 3Com IP Messaging module makes an outbound call to the reply phone number. When the called party answers, the 3Com IP Messaging module informs them of a message reply and prompts if they want to connect. If they select yes, then the mailbox owner and called party are connected. Once the called party disconnects from the call, the mailbox owner is placed back to the Message Review menu. The mailbox owner can disconnect from the live reply by pressing 9-9 during the live call.

For a reply with a message, the 3Com IP Messaging module allows the mailbox owner to record a reply message. The original message can be attached to the reply. The 3Com IP Messaging module sends the message by placing it in the senders mailbox or by dialing out to the sender.

Is there a way to warn a caller that is leaving a voicemail that they are approaching the maximum message length?

Subscriber Profile/Prof Max Record Time Warning specifies how many seconds before the end of the message to issue a warning that the max record time is approaching. The default is 10 seconds, so nothing should need to be done to make this happen.



2.11 Conferencing Requirements

What are the configuration options available with your conferencing solution?

The 3Com IP Conferencing solution supports three different configuration options:

“All-in-One” Configurationo Components installed on the single server:

conference server/conference attendant presence server VCX database conferencing and presence database (master) web console server

Dual Configurationo Components installed on the primary server:

conference server/conference attendant presence server VCX database conferencing and presence database (master) web console server

o Components installed on the secondary server: conference server/conference attendant conferencing and presence database (initial slave)

Distributed Configurationo Components installed on primary server:

presence server (if purchased) VCX database conferencing and presence database (initial master) web console server

o Components installed on secondary server: conferencing and presence database (initial slave)

o Components installed on all other servers: conference server/conference attendant



Describe how your conferencing solution can serve in a distributed architecture?

The 3Com IP Conferencing solution supports a distributed conferencing architecture that allows scalability and redundancy of the conferencing database.

Single server supports a maximum of 300 conference participants Distributed configuration supports up to 3,000 concurrent conference participants Single web console to set up and manage conferences All ports for any given conference is on the same server and if that server fails during

a conference, the conference is dropped Routing to conferences on any conference server can be performed by any

conference attendant or conference server in the pool– In a single server (“all in one”) implementation, provision the VCX to route to

the conference server and conference attendant components on that single server

– In a multi-server configuration, provision the VCX to route to conference server and conference attendant components on at least two servers

Database redundancy is provided to ensure no single point of failure in the distributed system

All of the servers in a distributed conferencing system must belong to the same domain


IP Phone Convergence Center Client

3Com VCX

Conference domain

Conferencing Conferencing Conferencing Conferencing

-Conferencing- Routing

-Slave DB-Routing

-Master DB-VCX DB-Web Console-Presence

Primary Secondary


Briefly describe your audio conferencing functionality.

The 3Com IP Audio Conferencing application provides an all-IP, SIP-based audio conferencing solution that runs on a hardened, Linux-based, industry standard enterprise server. The audio conferencing application supports scheduled and meet-me conferences for up to 300 users on a single server. Different server types are available depending on scalability requirements, including the IBM X-Series 306 and 346 servers.

Users can join audio conferences from the IP network or the PSTN network. From the IP network, 3Com SIP phones and the 3Com Convergence Center Client (soft phone) provide simple dialing and click-to-join functionality. From the PSTN network, 3Com’s media gateways provide access for users dialing in from their office, mobile phone, or home phone via a PSTN-based number to gain access to a conference attendant.

The audio conferencing application provides simple conference provisioning via standard web browsers, automated announcements for entry/exit/end of conference, and web-based conference control functionality. The 3Com IP Conferencing module supports the ability to define moderators, participants, and to support lecture-mode conferences. The 3Com IP Conferencing module also supports email notification of conference creation and modification.

The cost savings and resulting return on investment of the audio conferencing application can be significant. The audio conferencing solution turns into a vehicle to drive revenue for <<clientShort>> by implementing the application in-house and by using media gateways across the enterprise to provide local-number access for users.

Can external users access an audio conference from the PSTN?

Using a media gateway to provide connectivity to the PSTN, external callers can access an audio conference by dialing a number that is routed through the media gateway to the 3Com IP Conferencing IVR. The external user enters the conference ID number followed by a passcode (if provisioned by the conference creator).

Describe the ability of your system to support scheduled conferences.

Scheduled conferences are specifically provisioned in advance by the conference moderator using the web provisioning interface via standard web browsers, and are intended for formal conferences.

The maximum number of participants is defined along with the start time and duration of the conference. A scheduled conference “reserves” ports on the server for the maximum number of participants within the licensed number of users. Access is guaranteed for the configured number of participants between the scheduled start time and duration. If ports are available, additional attendees beyond the scheduled maximum may attend the conference. Users can attend a conference from a SIP device or from the conference attendant, which is an IVR application running on the 3Com IP Conferencing server. Conferences are identified by a “conference ID”, and can support public or restricted access to the conference.



For audio conferences, announcements or tones will sound when participants join and leave a conference, and when the end of a conference is approaching. These announcements can be configured by an administrator or conference creator/moderator.

Describe the parameters used when scheduling a conference.

A standard web browser is used to access the 3Com IP Conferencing administration interface. The user enters their VCX extension with domain, and clicks on Schedule Conference to begin.

General:

Enter numeric conference nameo Leave blank or select a number between 1000-1099. It is recommended this

is left blank to ensure conflicts with other conferences do not occur. Subject

o An alpha numeric string with no special characters like commas, periods, etc. Max number of participants

o Choose more than you think you need. Media type

o Defaults to audio, select video and desktop sharing if you think it might be used.

Schedule:

Start date and timeo Click on the calendar icon and select a start date and time. o The time will be in the time zone configured for the IP Conferencing server

during installation.o It is a good habit to schedule the call for 5 minutes before the actual start. If

this is not done, users will not be able to join the call until exactly the time you have scheduled.

Duration in minuteso Enter how many minutes the call will last. It must be between 15 and 1440

minutes (1 day). o The conference will be automatically terminated at the end of the duration,

dropping all participants and moderators, unless a moderator first extends the conference using the In Conference help system.

o The conference duration cannot be longer than the administrator-defined maximum duration. The administrator can over-ride this on a per-conference basis.

Access control:

Conference typeo This is asking if you want to use a passcode or not. o Restricted requires a passcode be entered in the next field below.o Public requires no passcode to join the call.

Participant passcode for restricted conference



o This is asking what you want your passcode to be. Moderator passcode for restricted conference

o This is a special passcode you can set for the moderator(s) of the call. Additional features are available to the moderator from the help menu (**).

o The system will not allow you to set a moderator passcode that is the same as the participant passcode.

Conference Announcements:

First in conference repeat intervalo Sets, in seconds, how often you would like the system to tell the first caller

they are the first caller. Join announcement type

o Select if you would like participants to be announced by name (like meeting place), a short tone, or no notification at all.

Leave announcement typeo Same as above except this is done when the participant leaves

End of conference first warningo Sets, in minutes, how long before the end of the meeting you would like the

system to give you a warning. Again, the meeting will disconnect all users when the scheduled time is up.

End of conference last warning o Same as above

Describe the ability of your system to provide meet-me conferences.

Meet Me conferences are setup in advance by users without a start time and duration, so they are always available for use, intended for informal conferences. Meet Me conferences can be setup for public or restricted access, and have the same capabilities as scheduled conferences.

Typically, a Meet Me conference is setup by a user to have a personal conference bridge. The system administrator must consider the capacity of the conference server to provide Meet Me conference capability in a reliable manner, as Meet Me conferences are only allowed when system resources are available.

For audio conferences, announcements or tones will sound when participants join and leave a conference. These announcements can be configured by an administrator or conference creator.

Describe the ability of your system to provide dial-out conferences.

Instant and emergency conferences are identical with the exception of the following:o Message Waiting Indicationo Users receive a MWI if they fail to join an emergency conference when

invited. MWI is not available with instant conferenceso Continuous Alert o Users can be alerted by an alarm or flashing light that an emergency

conference is taking place



o Continuous alerts are not available with instant conferences Configuration Authority

o Only administrators have the authority to configure emergency conferenceso Any user can configure an instant conference

Source Addresso You can use any valid user address as the source address (“From” address)

for emergency conferenceso Instant conferences always use the conference creator’s address as the

source address.

What is the maximum number of users that can be on a single conference call?

The 3Com IP Audio Conferencing application supports scheduled and meet-me conferences for up to 300 users on a single server, with up to 100 users on a single conference call. Multiple industry-standard enterprise-grade server types are available depending on scalability requirements.

In the current release, IP Conferencing servers are standalone units in that they cannot be aggregated to increase the number of users on a single conference. Conferences are constrained to a single server.

Is user data common between the conferencing service and the phone system? How is data imported to the conferencing service?

All user data and credentials used by the 3Com IP Conferencing Server and Presence Server originates from the VCX. The VCX data import application ensures the user database on the VCX is mirrored to the IP Conferencing Server and Presence Server. The primary server of a VCX redundant pair is used to obtain the data from the VCX.

The import file will periodically be retrieved from the VCX using secure copy (scp) and imported into the IP Conferencing and Presence database. The location of the original file and the period of retrieval is provisioned from the web interface by administrators. The import file does not contain clear text for user passwords. The import file contains an MD5 hash of user credentials, specifically H(A1) as defined in RFC 2617.

How does the conferencing application ensure access to all participants (internal and external) that have been scheduled?

The 3Com audio conferencing application supports public access and restricted access audio conferences. For public access, users need to know only the SIP URI or IVR extension when using a SIP device, conference ID when using the conference attendant.

For restricted access, users must either be on the access control list for the conference or know the conference ID and password when using the conference attendant. A DTMF passcode is utilized to prevent unauthorized entry into the conference. This passcode enables the moderator to login separately, giving them additional rights to administer the conference that is above and beyond the normal participants, such as the ability to put the conference in lecture mode.



The access control list is created manually by the conference creator or automatically by the conference attendant, and is reserved for those who will be moderators. The 3Com IP Conferencing module uses SIP digest authentication to verify users via SIP.

What privileges do regular users have to administer conferences?

Account information for regular users is imported to the 3Com IP Conferencing Module from the VCX. If user information (name, SIP address, etc.) for a regular user requires updating, this must be done on the VCX, not the 3Com IP Conferencing Module.

A regular user has the authority to do the following in the 3Com IP Conferencing Module:

Add their personal e-mail address Add Scheduled and Meet-me conferences View all Scheduled conferences View their own Meet-me conferences Manage their Presence Access Control List.

What privileges to system administrators have to schedule conferences?

On installation, a single super user is created. This super user cannot create other users. Users are imported from the VCX. The super user is purely local to the 3Com IP Conferencing Module, has administrative privileges that cannot be removed, and the super user cannot be deleted. The super user can assign administrative privileges to imported VCX users. All users with administrative privileges can assign or remove administrative privileges from all users except the super user.

Administrators can access the following options through the Admin menu of the 3Com IP Conferencing Module:

System Configuration — Contains parameters which control the operation of the entire system. Includes global configuration, conference server configuration, local domains, presence settings, XML database import settings, and licensing information.

User List — The complete list of user accounts. Administrators can select accounts from the list and review them in detail. They can also perform limited updates (change passwords, add e-mail addresses) to the accounts. The addition or deletion of users must be done on the VCX.

Monitor Servers — The Monitor Servers screen, where administrators can check, start, and stop system processes.



Describe how users can control a conference from a web browser interface.

You can use the 3Com IP Conferencing Module’s Conference List to view Scheduled and your own Meet Me conferences. Depending on your authority level, you can also delete conferences from the Conference List. If you are the conference owner, or if you have been assigned conference moderator privileges by the conference owner, you can delete the conference.

Conference properties can be viewed and edited. Any user can view properties for a public conference. However, only the conference creator or conference moderator can edit properties for a conference. The Conference Control screen is divided into two sections, Conference Control and Participant Control. The Conference Control section allows a conference moderator to modify the settings for the conference. The Participant Control section allows a conference moderator to change access settings for participants of the conference.



Can users be notified of a scheduled conference?

The 3Com IP Conferencing module provides e-mail notification to the conference creator, and allows all users to view the conference status using the IP Conferencing web-based administration interface.

Once all of the appropriate settings for a conference have been selected, click Submit. If successful, you should see the following message:

“The conference has been created. The email has been sent to [email protected]”

This means the conference has been created and it has sent the moderator an email confirmation to the email address that has been configured in the user profile. The moderator will receive an email from the IP Conference Server with the subject of “Conference Service Confirmation”. This confirmation will include the conference ID, the passcode for the participants, and the passcode for the moderator.

All users can also confirm that the conference has been created by selecting Conference List from the pull down or the link on the right side of the screen.

Can a roll call of participants currently in the conference be announced or displayed?

The 3Com IP Conferencing server allows moderators to hear a private roll call at any point in a conference. While in a conference, a moderator dials ** to access the In Conference help system, then enters 8 to hear a roll call. While the roll call is being announced, the moderator will not be able to hear the other participants, who will not be able to hear the roll call. The moderator presses 9 at any time to exist the roll call and return to the conference.

Any user who has a login for the IP Conferencing server can see a list of conference participants using the IP Conferencing web administration screens.

In addition, users accessing the conference through the 3Com Convergence Center Client can use the Instant Messaging box for a display of all participants currently on the conference. This is done by typing “who” in the Instant Messaging box while currently on a conference.

Describe the audio codecs that are supported by your audio conferencing solution.

The 3com IP Conferencing Module supports conference participants using any combination of the supported CODECs. The audio conferencing server trans-codes as required.

Supported CODECs include:

G.711A-law (pcma) G.711 Mu-law (pcmu) G.729 G.721 DVI ADPCM



GSM G722

Note that the 3Com Convergence Center Client supports G.711 Mu law and G.729 audio codecs.

What is the maximum number of users on a single audio conference?

The 3Com IP Audio Conferencing application supports scheduled and meet-me conferences for up to 300 users on a single server, with up to 100 users on a single conference call. Multiple industry-standard enterprise-grade server types are available depending on scalability requirements.

In the current release, IP Conferencing servers are standalone units in that they cannot be aggregated to increase the number of users on a single conference. Conferences are constrained to a single server.

Describe how users can control an audio conference using DTMF input.

Moderators can control active conferences using the dial pad on a phone, or on the 3Com Convergence Center Client. Individual participants in a conference can mute and un-mute their own voice.

Moderators can:

• Extend conferences (scheduled conferences only)• Mute and un-mute the voices of all participants in a conference (excluding all

moderators)• Mute and un-mute their own voice • Lock and un-lock a conference• Hear a private roll call of all the participants in a conference.

The In Conference Help system guides the user through the process of controlling an active conference using a dial pad. The system prompts the user to press the appropriate key. For example, “Please press 6 to mute or un-mute yourself”.

To start the In Conference Help system:

Using the dial pad, press **

The help system starts and presents you with a list of options. The functions you can perform using a dial pad depend on your status and the type of conference you have joined.

In Conference Help system controls include:

** Starts the help. *4 Extends the conference by 15 minutes. (only available to moderator) *5 Mutes and un-mutes the voices of all participants except conference moderators.

(only available to moderator) The conference moderator can mute the voices of all



participants in the conference. This is equivalent to a “lecture mode” as the moderator’s voice is the only voice that can be heard.

*6 Mutes and un-mutes your own voice. All conference participants can use this feature.

*7 Locks and unlocks a conference. All conference participants can choose to lock access to the conference. Once locked, no additional participants are allowed to join the conference, but active participants are free to leave.

*8 A roll call will sound listing all of the participants in a conference. Participants can use this feature to find out who is in the conference. When a roll call is in progress you will not be able to hear participants talking in a conference.

*9 Exits the help.

Does the system offer video conference capability inherent to the core product?

The 3Com Convergence Applications Suite includes video conferencing that supports the following features:

Works with H.263 video codecs Scheduled or ad hoc video conferences One-to-one, one-to-many Max. video sessions limited by CPU / memory 1 session/client, with 8 or more users



2.12 Presence

Describe your presence functionality.

The 3Com IP Telephony Suite includes an optional IP Presence feature that enables users to see the availability of colleagues. The presence functionality is based on SIP and avoids “blind” calling and enables more efficient communications.

The 3Com IP Presence Server supports the following features:

Telephony presence maintained by VCX Publisher: publish state to Presence module Subscribers subscribe to publisher’s state Forms ‘Contact lists’ On-hook/off-hook, on-line/off-line SIMPLE-based Instant Messaging

Describe the ways that your presence interacts with users.

Users interact with the IP Presence Server as follows:

Subscribe to monitor the status of their buddies Receive the presence status of their buddies Maintain the Access Control List that will be used by the IP Presence Server to

determine who has authorization to monitor someone’s presence Maintain the user preferences such as notification types, addresses, automatic

access approvals or denials, etc.)



2.13 Call Center

Does the proposed solution support an integrated inbound call center application?

Does the call center application require additional server hardware?

Does the call center solution provide real time and historical reports?

Does the call center solution provide support for skills-based routing?

Describe the capacities of the call center application.



2.14 System Administration

Describe the process for administrators to remotely access the system. Identify if any special software or plug-in is required.

The VCX is easy to manage by providing centralized access to all systems using web-based interfaces either locally on a LAN or remotely over a WAN by multiple administrators and end users (if allowed). The call control and messaging systems are managed from centralized servers that reach out to distributed locations to inter-connect the whole enterprise.

The VCX makes life easier for voice/IT staff by providing the ability to import bulk provisioning data from ASCII flat files, supporting regularly scheduled back-ups and reports, generating call detail records for offline analysis, and by providing real time tools for system status, problem identification and resolution, and system utilization.

A single workstation can administer multiple remote sites. Additionally, using 3Com’s Web Provisioning service, which is included with the VCX, changes to a single database (for subscriber information, dial plans, etc) can be automatically distributed to remote VCX that are part of the system.

The VCX system provides a centralized console manager to manage multiple systems through the same web interface. All systems have the same functionality whether monitored remotely or locally.

The equipment required to perform administration is a standard desktop computer running a standard web browser (Internet Explorer is currently the qualified browser). There is no other additional hardware or software required to administer branch offices and remote sites.



Does your solution provide central administration and management of all users and equipment across all locations?

Basic system administration is performed through a standard web browser interface. A single workstation can administer multiple remote sites. Additionally, using 3Com’s Web Provisioning service, which is included with the IP Telephony solution, changes to a single database (for subscriber information, dial plans, etc) can be automatically distributed to remote IP Telephony nodes that are part of the system.

The System i IP Telephony solution provides the ability to import bulk provisioning data from ASCII flat files, supporting regularly scheduled back-ups and reports, generating call detail records for offline analysis, and by providing real time tools for system status, problem identification and resolution, and system utilization.



Can multiple administrative activities be performed by multiple administrators concurrently?

Outline the steps required to facilitate setting up a new extension.

Does the proposed solution support the ability to map feature functions to phone buttons on a per-user and per-group basis?

Outline the steps required to move a phone between buildings.

Is your system compliant with government standards on time zones?

The 3Com VCX IP Telephony solution supports multiple Time Zones in the current release.

The time zone is assigned on a per-extension basis by administrators from the VCX Administration web provisioning interface, or by users from the VCX VoIP User web provisioning interface.

Administrators and users can define the following time zone settings on a per-extension basis:

The date/time format used for the 3Com IP Phone displays Time zone Observe daylights savings time (yes/no)

An example of the VCX VoIP User web provisioning interface is shown in the following screen shot.



Does the proposed solution support automated daylight savings time?

The VCX supports automated daylight savings time settings within time zones that support daylight savings time.

Could there be several security levels established in order to access the administration applications?

New user types of Administrator, manager, dir added to increase tiers of administrationo The manager role allows access to menu options that manage users and phone

extensionso The dir role allows access to menu options that manage routing services

More control and administrative levels Increased security

Does equipment support line command administration?

Does the administrative application have on-line help?

Does the system permit the system administrator to locate station information based on multiple criteria (e.g. extension number, name etc.).

Does the system support templates, which allows the system administrator to program multiple telephones with similar features/functions at the same time?

How frequently does the bidders call processing system back-up the configuration data, which includes up-to-date moves and changes?

Is the system capable of doing such a backup remotely to a secured off-site without on-site administrator presence?

Describe the Describe Class of Service restriction levels available to define calling patterns for telephones.

Describe the ability to restrict the features available to end users via the administration interface.

Identify the languages supported by the administration interface.



What normal maintenance/ administrative activities require system downtime? List the system down time incurred by each of these activities.

What type of system maintenance do you suggest for the system? Do these procedures require downtime on the system? If so, how long?

Describe the US 508/ADA Compliance of your system.

The 3Com IP Telephony Suite supports a web provisioning interface for end users and administrators that is US 508/ADA compliant.

As a matter of background:

Section 504 - Section 504 states that "no qualified individual with a disability in the United States shall be excluded from, denied the benefits of, or be subjected to discrimination under" any program or activity that either receives Federal financial assistance or is conducted by any Executive agency or the United States Postal Service. Each Federal agency has its own set of section 504 regulations that apply to its own programs. Agencies that provide Federal financial assistance also have section 504 regulations covering entities that receive Federal aid.

Requirements common to these regulations include reasonable accommodation for employees with disabilities; program accessibility; effective communication with people who have hearing or vision disabilities; and accessible new construction and alterations. Each agency is responsible for enforcing its own regulations.

Section 508 - Section 508 establishes requirements for electronic and information technology developed, maintained, procured, or used by the Federal government. Section 508 requires Federal electronic and information technology to be accessible to people with disabilities, including employees and members of the public. An accessible information technology system is one that can be operated in a variety of ways and does not rely on a single sense or ability of the user. For example, a system that provides output only in visual format may not be accessible to people with visual impairments and a system that provides output only in audio format may not be accessible to people who are deaf or hard of hearing. Some individuals with disabilities may need accessibility-related software or peripheral devices in order to use systems that comply with Section 508.

There are many components to ADA. The two biggest issues being accessibility and discrimination. The major requirements in ADA compliance are telephones being Hearing Aid Compatible (HAC), having features designed to help the visually and hearing impaired and TTY compliance.

The VCX is compliant with ADA for the user provisioning interface. Additional levels of compliance will be achieved in Roadmap releases of the software. 3Com also works to makes its equipment compatible, where it is readily achievable, with peripheral devices used by people with disabilities.



2.15 Reporting

Provide a brief overview of your system’s data reporting and real-time monitoring capabilities.

The 3Com VCX solution includes components that are responsible for accounting and Call Detail Record (CDR) functionality. CDRs contain call specific information that is used for billing, monitoring and analyzing traffic, and troubleshooting call failures. The VCX Accounting Data Service allows the Call Record Service to retrieve and delete accounting records. The VCX supports Call Detail Records for single site and multi site implementations.

Can your system aggregate performance data from multiple PBX systems, sites, servers, and/or components?

The Accounting Service stores Call Detail Records (CDRs) generated by the VCX call elements for both successful completed calls and unsuccessful call attempts. Once the call is received, CDRs are created for that call (whether it is successful or not). The data is then forwarded to the Accounting Server. The architecture of the VCX CDR collection and reporting system is illustrated below.

In this architecture, there is an Accounting Service running with each VCX Call Processor, and one global Call Record Service. The Call Record Server is global across all branches and all regions across the enterprise (i.e. there is one and only one Call Record Server across all VCX regional offices within an enterprise). All VCX CDR files are flat ASCII files in ASN encoded XML format. Each field in a CDR has its own abbreviated tag name and an


Call Record(CDR) Server

Secure FTP

CDRReporting

Application

Super-CDRASN Encoded

XML Files

Merge Rules

Connection Rulesfor pulling XML files

from each Accounting Server

Regional Office IPT(Primary)

Regional Office IPT(Secondary)

Branch Offices

SIP CallProcessor

AccountingServer

ASN EncodedXML Files

SIP CallProcessor

AccountingServer

ASN EncodedXML Files

Regional Offices


associated value. The abbreviated tag name to actual field name mapping is maintained at the Call Record Service. The originating Call Processor creates a CDR with a unique text call identifier, inbound device type, and outbound device, which are propagated to each Call Processor during the course of a call.

Because CDR requests may arrive at high rate, the Accounting Service periodically starts new XML files to prevent the files from growing beyond a manageable size. Other algorithms such as maximum file size, maximum number of CDRs per file also may be used for this purpose, which is chosen by a system administrator using a configuration file. The Accounting Service automatically deletes on a preset basis after the files have been downloaded by the Call Record Service.

The Call Record Service queries a list of Accounting Services on a periodic basis, reads the files, and merges the CDRs into Super CDRs according to preset rules. The XML files are pulled from the Accounting Services by means of any modern transport protocol, such as secure FTP, HTTPS, or file sharing. The Call Record Service has a corresponding resource connector for each protocol and customers specify the preferred protocol, group membership, and frequency of retrieving CDR files using an XML configuration file.

Once a CDR file is downloaded from an Accounting Service, it is placed in the Inbox folder for its server group. The Merge/Export task may be scheduled to run on a periodic basis or to be performed on demand. Each Inbox folder has its own transformation rules and those rules can be changed by administrators at any time. The rules are expressed in a form of an XLST document. This is a standard language for XML processing and transformation. The connections between the Call Record Service and each Accounting Service are connected and maintained only for the duration of each download.

Describe the historical reports that can be generated from the call detail record system of your solution.

The 3Com VCX CDR Reporting application is a Windows-based application that administrators use to analyze call reports based on extensions, destinations, and messages. The 3Com CDR Reporting tool aggregates information from the billing support services and generates pre-canned reports. The VCX CDR Reporting application provides a modern GUI look and feel with advanced data viewing, sorting, and searching functions. This application retrieves CDR data directly from the VCX IP Telephony Call Record Service and the IP Messaging CDR Service using secure FTP on an on-demand or automatic basis.

The VCX CDR Reporting application supports up to 250,000 call records in a single XML file. Pre-canned reports include:

Incoming call report Outgoing call report All call report All call report per extension Calls per hour Internal call report Reports on the messaging system Reports by branches Hunt group reports:



o Average call duration for hunt groupo Call Distribution for hunt group by agent by dayo Call Distribution for hunt group by agent by houro Call Distribution for hunt group by dayo Call Distribution for hunt group by hour

Monitor Barge-In



Describe how data is viewed for historical reporting.

The VCX CDR Reporting application provides an easy to use data grid to view IP Telephony and IP Messaging CDR details, and perform sort, search, filter, export, and print functions.

The VCX CDR Reporting application supports these Data Viewing features:

Call Data grid to be based on the user’s Window system colors/theme look and feel Grid performance enhanced through “virtual mode”. This mode supports enormous

amounts of data, allowing fast binding and scrolling through literally millions of rows In the Call Data Grid, the user can:

o Specify the order of the call data fields displayed in the call data grido Specify which call data fields are to be included in the call data grido Sort the call data grid in ascending or descending order o Specify filter (e.g. query) criteria on one more fields in the call data grido Specify columns as pinned (non scrolling); allows these columns to remain in

sight as the user horizontally right scrolls the CDR call data grid Allow the user to save all the grid configurable information as a new grid view for

subsequent reuse Export the contents of the call data grid to Microsoft Excel Print the contents of the call data grid

An example of VCX IP Telephony data in the Data Grid is shown below.



An example of IP Messaging data in the Data Grid is shown below.



The VCX Call Reporting application allows users to view the data they way they want to see it. Within a Grid data view, users simply select a column and drag it left or right to move it to a different column position. Many different views can be created and easily navigated to by saving the view as shown below, and clicking on the view name from the Grids menu.



In addition, users can sort the data view in ascending or descending order by any column by clicking on the column heading.

Describe the licensing requirements for your historical (call detail record) reporting system.

There is no need to purchase a license for Crystal reports if the pre-canned reports that come with VCX are sufficient. Crystal Reports (version 9 or above) is only needed if extra or advanced reports have to be generated.

Describe the server requirements for your historical (call detail record) reporting system.

This is a Windows application that runs on customer-provided server, PC’s, or laptop with the following minimum specifications: Windows XP, 1 GB RAM, 30 GB Hard Disk.



Does your system provide the user with the ability to create custom real-time and historical reports?

3Com provides CDR reports in the following formats:

PDF Text Tab-separated text Crystal Reports Excel Word Rich Text Format XML Record Style (columns no spaces) Record Style (columns with spaces) Report Definition Separated values (CSV) Rich Text Formatted

In addition to the pre-canned reports, the administrator can also create their own reports using this report generating tool. The VCX CDR Reporting Tool also provides time-saving report querying capabilities, including:

Enhanced report querying o Search called and/or calling party with wildcards and save the search for

subsequent re-use Dynamic query/printing grid content

o Allow users to filter the records that appear in the grid with a query dialog and allow the grid contents to be printed

Saving and retrieving queries o Allow users to save, retrieve, update, and delete queries

The raw data downloaded from the VCX IP Telephony and IP Messaging servers can be saved to a comma separated value (CSV) file.



Provide examples of the standards reports available with the proposed solution for the telephony and voice mail sub-systems.

The reports that are available with the VCX Call Reporting application for IP Telephony are shown in the illustration below.

To generate a report, click on the report name in the Reports menu. The following dialogue box will appear prompting you for some report information. The report period can be today, this week, last week, this month, last month, this year, last year, and custom. The report can be filtered for any one calling or called party ID, and the report can be sorted by a particular column.



Custom reports can be created and retrieved using the VCX Call Reporting application. Enter a title and click Save to create a custom report. Generate a report from a saved custom report by clicking on Retrieve. An example of an All Calls report is shown below.

An example of an Average Call Duration By Hunt Group is shown below.



An example of a Malicious Call Trace report is shown below.

An example of a Gateway Utilization report is shown below.



An example of a Call Distribution for Hunt Group by Agent by Day report is shown below.



The VCX Call Reporting application also supports graphical reports, as shown in the Calls per Hour report shown below.



The IP Messaging reports that are available with the VCX Call Reporting application are shown in the illustration below.

An example of a Voice Mail report is shown below.



2.16 System Management

Briefly describe the network management capabilities of the proposed solution.

As an optional component of the IP Convergence Applications Suite, 3Com’s Enterprise Management Suite (EMS) provides tools that enhance the manageability of the IP Telephony and IP Messaging platforms. The EMS is a highly scalable, multi-user, client/server, object-oriented network management system that abstracts service logic from individual element management. The EMS uses a data driven model, as network inventory, device attributes, management methods, bulk operations, scheduler, users, and events are stored in the same database. Based on open standards, the EMS supports the ability to integrate with other management tools, such as HPOV, NNM, and Tivoli.

As shown below, the EMS client provides an organized view into the voice (and data) network infrastructure, using right clicks to hone-in on context-sensitive attributes and operations.

The management operations that can be performed using 3Com’s EMS application include:

Creation and use of logical views for IP Telephony servers and gateways System security Monitoring system health SNMP Traps System Configuration Planned Software Upgrades Configuration and User Data Backup Application-specific management and operations



Media Gateways Integration available with EMS for IP Convergence Applications:

o Using scheduled operations for VCX upgradeso Using data collector to collect call statisticso Using data collector to collect channel stats on gatewayso Using templates to configure parameters on gateways

Describe the server and database requirements of your management solution.

The requirements for the EMS server include:

Customer-provided Windows 2003, Linux, or Sun server Pentium 4 processor or similar 2 GB RAM 533 Mhz Front Side Bus 30 GB Hard Disk

The database requirements for the EMS server include:

Product ships with a mysql database which is good for very small or trial systems Product requires an SQL or Oracle database for normal enterprise-grade operation

The requirements for the EMS client include:

Windows XP Pentium 4 processor or similar

Can the system be programmed to perform system backups automatically?

Automated and scheduled backup functionality is provided with EMS for all VCX configuration information. User Information, Dial Plan, etc are backed up separately from the Command-Line

EMS (Enterprise Management Suite) provides full VCX system backups of all configuration information. User information is handled via the Oracle Backup functions from the Command-Line. Backups can also be scheduled for convenient and appropriate times. The backup/restore feature uses the Common Agent MIB and sees the backup file format changes from ‘CFM’ files to UNIX TAR files. All Backup and Restore functions can be performed across the LAN and WAN. Backup of a fully configured VCX takes approximately 120 seconds, the time may vary depending on network utilization and the processing capability of the management system.

Does the management system support the graceful shutdown of services?

Graceful shutdown is supported for the IP Telephony and IP Messaging components. Active calls are not terminated. Upon shutdown, all the end points (phones/gateways etc) automatically move to the secondary so that there is no disruption of calls.

Describe the interoperability with other OSS systems.

The 3Com solution is fully SNMP compliant and may be integrated with open standards based network management suites such as HP Openview. The 3Com solution also



provides a native management application, the Element Management System (EMS). The EMS provides rich, graphical, element status and health views, call and messaging activity, and customizable logging views.

Can the management system support the ability to perform scheduled software upgrades?

Software upgrades are performed by taking one of the redundant call processing servers offline (causing the other server to be primary for all calls). The offline server is then upgraded from CD or from a CD image that has been downloaded from 3Com’s FTP server. When this upgrade is complete (typically within one hour), the server is brought back online and the other server is taken offline for upgrade. Software upgrades can be performed by 3Com, partner-trained, or customer-trained personnel.

A small maintenance window is preferred during upgrades. Individual servers can be upgraded with the system being live as long as there is a redundant component. However, the IP Telephony server needs a small window to synchronize its database. During this period of synchronization, the system will continue to operate by directly making calls via the gateways but go into a limited functionality mode.

Describe how we can monitor the status of equipment in your proposed solution. Does the equipment send SNMP alerts?

There is extensive monitoring capabilities with a fully integrated EMS management system with no impact on service. The 3Com VCX solution uses SNMP to traps to provide event notification of software components. One function of the 3Com Enterprise Management Suite is to provide the ability to monitor VCX system events, and to alarm and notify as configured by the customer.

In addition, the 3Com Enterprise Management Suite can provide many other network management capabilities such as monitoring the health of VCX software processes, providing reports, and managing the configuration of all components.

The VCX solution also provides log files that can be viewed in real-time or examined at a later time for component failures.

Proactive management is provided in the large number of traps (unsolicited alerts) that the VCX is capable of generating. Additionally, by setting thresholds on suspected variables further diagnostics can be performed to isolate and problem.

EMS can generate a On-the-fly report of any counter based variable by simply selecting the variable from the UI and selecting ‘Collect Now’. Historical information collected by data collectors can be reported as well. Historical data can be viewed in a number of formats as well as time segments.

No additional components are required to run reports.

The 3Com VCX solution uses SNMP to traps to provide event notification of software components. One function of the 3Com Enterprise Management Suite is to provide the



ability to monitor VCX system events, and to alarm and notify as configured by the customer.

3Com VCX system runs self diagnostic process that automatically restarts any job/processes that become inoperable. In addition alarm notifications are provided through SNMP traps to the 3Com EMS. Alarm notifications are also logged in log files in the systems for various non critical errors.

Can QoS monitoring be done through SNMP for monitoring with your network management solution?

EMS is the 3Com designated tool for voice call quality monitoring. EMS provides the ability to capture, store, and report on voice statistics as well as set thresholds for proactive notification.

3rd party applications can take advantage to the VCX instrumentation and provide monitoring as well utilizing the 3Com MIBS.



2.17 Implementation

Fully describe the implementation process.

3Com adheres to a five (5) phase approach to Project Management, covering Sales, Planning, Implementation, Testing and Servicing.

The Proposal stage provides the foundation for a successful project. A key element during this phase is to clearly understand the 3Com commitment needed to meet the Customer expectations. Other critical elements during this phase are:

Project team creation Customer requirements reviewed and confirmed Technical solution development Proposal and supporting documentation development

Upon successfully closing the sales process, the Planning phase indicates the first step in implementing the technical solution. During this phase the 3Com Project team will begin a series of Customer meetings that:

Finalize the technical requirements Assess the existing environment Detail responsibilities – 3Com/Customer Review the 3Com planning documents Customer Information document (CID) System Planning Guide (SPG) Statement of Work (SOW) Define the installation timeframe Develop the Project Plan


Phase I

Proposal

Phase II

Plan

Phase IV

Test

Phase III

Deploy

Phase V

Review

Sales Implement

PROJECT MANAGEMENT

Support

Service and Design

Monitoring

Phase I

Proposal

Phase II

Plan

Phase IV

Test

Phase III

Deploy

Phase V

Review

Sales Implement

PROJECT MANAGEMENT

Support

Service and Design

Monitoring


In accordance with the developed implementation timeline, the Deployment stage consists of the following activities:

Equipment Racked & Powered Server Setup Configuration & Provisioning Dir Server - Configured Auth Server – Configured Installation of Gateways Installation & Programming of Users Perform - Functional System Test Perform - Acceptance Testing Cutover Users System - Live & Accepted

The object of the Testing Phase is to confirm that the 3Com Solution / Installation has both satisfied the Customer’s originally agreed requirements, and has met the technical specification defined by the design. The key elements of the Testing Phase are:

Test Plan finalized Cut-Over Plan agreed and issued in timely manner Cut-over conducted in accordance with agreed Plan Acceptance Tests carried out in accordance with agreed Plan Acceptance Test anomalies resolved Post Sales Support arrangements are initiated Solution Handed over to, and accepted by, Customers’ Operations Project documentation finalized Project Time Schedule and Risk Management Plan updated Stage / Site Close out Meeting held Any Follow-on Actions identified

The object of the Project Review is to assess whether the Project’s objectives have been achieved and the Customer’s requirements have been met. The key elements of the Post Project Review are:

Customer’s Project related complaints are formally recognized and processed Customer has accepted and signed-off the Solution / Installation Customer Post Project Review Meetings planned and conducted 3Com Post Project Review Meetings planned and conducted Observations gathered from both Customer and 3Com Post Project Review

Meetings, are formally recognized and processed Post Project Reports compiled and issued Review results recorded with details of follow-up actions and plans for their

completion Follow-up actions are expedited and completed.



2.18 Maintenance

Fully describe the maintenance plans available for the proposed solution.

3Com provides the following Maintenance programs:

Guardian 8x5xNBD

Complete on-site package for customers requiring a technician dispatched to their premises to assist with troubleshooting or to perform parts replacement. Includes on-site maintenance and unlimited telephone support from 6:00 am to 5:00 pm Pacific Time, Monday through Friday, software upgrades, and hardware replacement by the next business day. Some software features and releases that 3Com charges for separately are not included. Guardian 24x7x4

Complete on-site package for customers requiring a technician dispatched to their premises to assist with troubleshooting or to perform parts replacement. Includes on-site maintenance and unlimited telephone support 24 hours per day, 7 days per week including holidays, software upgrades, and hardware replacement within 4 hours. Some software features and releases that 3Com charges for separately are not included.

Express 8x5xNBD

Complete remote package for customers requiring advance hardware replacement, but who do not require an on-site 3Com engineer. Includes unlimited telephone support from 6:00 am to 5:00 pm Pacific Time, Monday through Friday, software upgrades, and hardware replacement by the next business day. Some software features and releases that 3Com charges for separately are not included. Express 24x7xNBD

Complete remote package for customers requiring advance hardware replacement, but who do not require an on-site 3Com engineer. Includes unlimited telephone support 24 hours per day, 7 days per week including holidays, software upgrades, and hardware replacement by the next business day. Some software features and releases that 3Com charges for separately are not included. Express 24x7x4

Complete remote package for customers requiring advance hardware replacement, but who do not require an on-site 3Com engineer. Includes unlimited telephone support 24 hours per day, 7 days per week including holidays, software upgrades, and hardware replacement within 4 hours. Some software features and releases that 3Com charges for separately are not included.



2.19 Training

Provide a description of the System Administration and Maintenance training courses.

In the IP Telephony VCX VoIP 5.x Systems Hardware and Software Overview course, students learn gain a working knowledge of the components of the VoIP 5.x system. How to place them in a working network, as well as the basic hardware and software required to successfully plan a 3Com system, using SIP Proxy applications and VCX interoperability.

This is an overview class of the components of hardware and software that make up the VCX system:

Basic information of the System Placement, Configuration and Operation.

VCX™ V7210 IP Call Processor Web Provisioning Server Directory Server Authentication Server Call Processor

VCX™ V7220 Accounting Suite Accounting Server Billing Support Server

Media Gateways V7111 Analog Gateway V7122 T1/Pri gateway SNMP and Web Management

VCX™ V7350 Unified Messaging System setup and administration User setup and administration

Training may be either on-site or in the 3Com facilities in Rolling Meadows, Illinois.



3 Vendor Overview

Describe your company’s experience with implementing SIP-based communications solutions.

3Com has had a major impact on computer networking since the creation of Ethernet in 1979 with over 1,300 issued U.S. patents and more than 300 pending U.S. patent applications to date. A leading provider of secure, converged voice and data networking solutions that reduce network complexity and cost for businesses of all sizes, 3Com has an annual revenue of $651 million (FY05 year ended June 3, 2005) and a total of 1,500 employees (FY06) in over 41 countries. 3Com’s corporate headquarters is located in Marlborough, MA.

3Com has been shipping IP Telephony solutions to the small to mid-size market since 1998 and has over 30,000 IP Telephony installations in over 30 countries worldwide. The 3Com Convergence Applications Suite, with roots in the carrier market since 1998, was introduced in 2003 as the first enterprise-class native SIP-based set of communications applications that now includes IP Telephony, IP Messaging, IP Conferencing, IP Presence, and IP Tele Commuter. This is just one component of 3Com’s full portfolio of enterprise-class switching, routing, wireless, and security products.

Since 3Com's founding in 1979 and creation of the Ethernet standard more than 30 years ago, the world has embraced 3Com’s vision of pervasive networking:

Every personal computer contains a network connection; Businesses are fundamentally built around a flow of information carried by a network,

and; Enterprises large and small are increasingly adopting "converged" networks that

include Internet Protocol (IP) telephony technologies to achieve significant cost savings and dramatic new functionality and features that enhance the bottom-line.

Today, under the leadership of President and CEO Edgar Masri, 3Com is focused exclusively on serving the enterprise data and voice networking market and has a strong balance sheet, renowned brand, large intellectual property portfolio and global presence. The company has world-class strategic partners and one of the broadest product lines and distribution channels in the industry. 3Com continues to define the way networks are built through superior engineering and by leveraging standards to reduce complexity, unlocking the hold of proprietary systems and lowering cost of ownership.

3Com distinguishes itself in the following fundamental areas:

Quality — leading technology innovation, service, distribution and reliability backed by a sound balance sheet;

Breadth — offering among the broadest array of feature-rich, competitive networking solutions;

Value — excelling at price to buy and cost to own - a superior cost structure combined with a focus on reducing complexity, and;



Innovation — creating disruptive technologies to drive adoption of converged networking (e.g. VoIP/IP telephony). 3Com's intellectual property portfolio includes 1,057 U.S. patents with 720 more pending applications.

Provide a brief history of the proposed solution.

3Com’s VCX IP Telephony solution is based on a 3Com-developed Voice over IP architecture that is currently powering some of the largest carrier networks in the world, including AT&T, MCI/WorldCom, China Unicom, and others.

To date, 3Com VoIP solutions have transported more than 20 billion minutes of billable voice traffic for these carrier customers. This same technology has been scaled down for use in enterprise networks, without sacrificing any of the reliability and quality parameters.

3Com’s IP Telephony Market Momentum:

1998 – First to market with IP-PBX (NBX)1999 – First distributed Softswitch framework1999 – Softswitch deployed at AT&T2001 – Industry’s first SIP-based Softswitch2001 – SIP Softswitch at MCI2003 – Introduction of VCX IP-PBX for multi-site medium to large enterprises2003 – First with SIP based convergence applications suite2004 – 16,000+ NBX systems shipped2004 – VCX Release 5 available 2005 – VCX Release 6 available2006 – VCX Release 7 available

With a legacy of more than 7 years of VoIP deployments, 3Com’s carrier-grade Softswitch technology is currently installed in service provider networks worldwide. The first iteration, v1.0, was deployed in AT&T’s residential long distance bypass network in 1998 providing VoIP transport in an application called “transparent trunking”. This technology allowed AT&T to bypass their CLASS 4 (tandem) switches and, more importantly, pass the call to the egress Local Exchange Carrier as an untariffed data PRI instead of a tariffed voice call.


Transport Network

CalledParty

CallingParty

TC1000

Local Network Local Network

LEC LECTandem Switch

IPBackbone

TandemNetwork

Tandem Switch

Tandem Switch

TC1000

GK DIR/PROV AS/BSS3Com 3Com

SoftswitchSoftswitch


The second revision of the VCX was developed primarily as a “Hosted Business Service” or “IP Centrex” offering for MCI/WorldCom’s Enterprise Connection solution. As 3Com added more and more user-facing features, the product began to look more like an IP PBX than a carrier trunking solution. This led to 3Com’s decision to offer this product directly to enterprise customers.



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