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Convergence Applications Suite

RFP Template Answers


Release 7.1

June 8, 2007
Version 1

http://www.3com.com
06/07
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 Corporation – SOW v.3_050604 Page 2 VCX™ V7000 Implementation


IP Telephony Request for Proposal Template
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Table Of Contents

1 General Information and Instructions....................................................... ..........................5


1.1 Company Profile...................................................................................................... .....5
1.2 Purpose of Project................................................................................................. .......5
1.3 Goals and Objectives.............................................................................. .....................6
1.4 Current Telephony Environment................................................................. ..................6
1.5 Current Data Environment...................................................................................... ......6
1.6 RFP Coordinator................................................................................ ..........................6
1.7 RFP Schedule........................................................................................... ...................7
1.8 Proposal Submission.................................................................................................. ..7
1.9 Vendor Site Tours........................................................................................... ..............7
1.10 Proposal Questions.................................................................................... ...............7
1.11 RFP Evaluation Factors................................................................................... ..........8
2 Requirements................................................................................................................. .....9
System Capacities............................................................................. ..............................9
2.1 System Architecture........................................................................ ...........................10
2.2 Redundancy and Reliability........................................................................... .............25
2.3 Scalability..................................................................................................... ..............28
2.4 Interoperability....................................................................................................... .....30
2.5 Data Network................................................................................ .............................33
2.6 IP Phones............................................................................................................... ....36
System Features............................................................................................ .................44
2.7 Calling Features....................................................................................... ..................48
Unified Messaging...................................................................................................... .....88
Conferencing Requirements.................................................................................... ......109
2.8 Presence................................................................................................... ...............120
2.9 Call Center......................................................................................... ......................121
System Administration.......................................................................................... .........122
2.10 Reporting............................................................................................................. ...127
2.11 System Management....................................................................................... .......141
Implementation ......................................................................................................... ....145
Maintenance................................................................................................................ ..147
Training................................................................................................. ........................148
3 Vendor Overview.................................................................................................... .........149
4 Pricing............................................................................................................... ..............152

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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

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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.

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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.

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• 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.

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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

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

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

Location Type 3-year growth Feature PSTN


% Survivability? Survivability?

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

Location Basic Business Manager Attendant WiFi


Phones Phones Phones Consoles Phones

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

Location Soft Phones with Standalone Software Attendant


Companion IP Phone Soft Phones Console Positions

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

Location Digital Analog PFTS


Trunks Lines Lines

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2.1 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.
Telephony Applications
Internet
IP
Tele Commuter IP IP
Messaging Conferencing
Administration
& Management IP Telephony

IP
IP Presence
Contact Center
Enterprise
Management Suite, SIP
Web Provisioning,
& CDR Reporting Media Gateways & SIP Devices
FXO and FXS Analog Digital Media Gateways SIP IP & Wi-Fi Phones Soft Phone
Media Gateways

PSTN

Analog
Phone and Fax

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 users and types of

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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.

HQ (Regional Office) VCX IP Telephony VCX IP Telephony


and IP Messaging and IP Messaging
(Primary) (Secondary)
Admin & Wrb IP Phones Intelligent
Provisioning, Mirroring
CDR Reporting, CAS EMS and TTS

Analog FXS
Replication Media
Gateway
Digital
Media Gateway Enterprise IP Network

Fax Analog
Analog FXO
Machines Phones
Media Gateway WAN
Replication
for IP Telephony
POTS (to Primary)
PRI
Branch Offices
POTS
PSTN FXS GW

PRI V6000
Dig GW LAN

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.

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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.

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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.

Session Initiation Protocol

Network Station
Gateways Gateways Conferencing

PSTN Analog Mobility


Phones
Private Unified
Networks Digital
Phones Messaging

Contact
MGCP IP Phones Center

H.323 Video Mail

TDM PCs Voice


Networks End Points Applications

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.

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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

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• 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

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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.

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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

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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.

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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.

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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.

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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).

HotSpot
Corporate User
Contact Center Network
Home
Internet User
NAT
Conferencing VCX
Traveling
IP User
Telecommuting Hotel
Presence Router
Module

Messaging

DMZ DMZ +Direct Standalone

Telecommuting Telecommuting Telecommuting


Module Module Module

Internet Internet Internet

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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.

Centralized Administration and Management


with Global Directory Services

Regional Office 1 Regional Office n


PSTN

WAN

Branch Office Branch Office Branch Office Branch Office


1-1 1-n n-1 n-1

PSTN

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.

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

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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.

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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.

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2.2 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.

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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.

Primary Site Secondary Site


Intelligent Mirroring
IP Messaging IP Messaging

IP Telephony Replication IP Telephony

LAN LAN

Replication

Branch Offices
WAN
LAN
Satellite Offices

LAN

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

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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 PRIMARY SECONDARY


solution supports an inherent redundant capability.
One of the call processors is identified as Primary
Real Time Replication
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
PRIMARY SECONDARY processor. The failover to the Secondary call
processor occurs automatically the next time an
Real Time Replication
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 PRIMARY SECONDARY


is multiple point of failure. Phones go to gateways
directly if the call processors fail, going into a limited Real Time Replication
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
PRIMARY SECONDARY routes. The IP Telephony call processor use
alternate routes to other gateways if a route to a
Real Time Replication 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.

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2.3 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.

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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.

All-In-One All-In-One
CDR Server -> IP Telephony + IP Messaging
Regional Office

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.

IP Telephony
CDR Server IP Telephony

Regional Office

IP Messaging IP Messaging

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.

Regional Office
IP
Auth & Dir IP Auth & Dir Messaging
Telephony
MMU MMU
CallP
CallP
MSU MSU
CDR Server

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

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(each regional office is provisioned separately but accessible centrally), all in a secure reliable
manner.

2.4 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.

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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 Availability
G.711 Fully Supported
G.726 Supported by gateways
G.728 Supported by gateways
G.729 Supported (except for TTS)
G.729a Supported (except for TTS)
H.323 V2 Supported via 3rd party gateway
Q.931 Fully Supported by Digital Media Gateway
802.1d Fully Supported
802.1p (LAN Prioritization) Fully Supported
802.1q (VLAN) Fully Supported
802.3af (PoE) Fully Supported
SIP RFC 3261 Fully Supported
SNMP v2 Fully Supported
FAX - Group 3 Fully Supported by gateways
FAX - Group 4 Fully Supported by gateways
T.38 Fully Supported
IP Precedence Fully Supported
DiffServ Fully Supported
Weighted Fair Queuing Fully Supported
RED Fully Supported
Weighted RED Fully Supported
RTP Fully Supported
RTCP Fully Supported
Policy Based Routing Fully Supported
IPv6 OS supports; full support on Roadmap
H.263 Fully Supported
TCP/IP Fully Supported
UDP/IP Fully Supported
DHCP 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?

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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.

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2.5 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.

IP Messaging IP Conferencing/Presence
VCX IP Telephony Servers Servers
Servers

Analog Media
Core VLAN for Servers Voice VLAN for Gateways
RTP Devices
Digital Media
Gateways

Core
Switches
Edge PoE Switches At The Core:
Core Routers
• Separate interfaces for traffic
Two Trunked VLANS between • Core VLAN for servers
Edge switches and Core switches/routers: WAN • Voice VLAN for RTP Devices
• Default Untagged VLAN (for PC’s)
• Tagged Voice VLAN (for IP phones)
Edge Router

At The Edge: Voice VLAN


• Switch Ports are configured for Edge PoE Switches
default VLAN and Tagged Voice VLAN

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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 Processor
o 3Com IP Phones
o 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

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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

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2.6 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)

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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 button
o Redial
o Conference
o Transfer
o Hold
• 6 fixed feature keys
o Volume up
o Volume down
o Mute
o Hands Free
o Forward to Voice Mail
o 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

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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 Redial
o Conference
o Transfer
o Hold
• 6 fixed feature keys
o Volume up
o Volume down
o Mute
o Hands Free
o Forward to Voice Mail
o 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

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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

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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.

>Presence
Availability >Instant Messaging
Controls
>Voice
>Video
>Desktop Collaboration
Presence &
Call >Voicemail View and Access
>Contact List
>Call history
>Import from
CSV — Outgoing calls
>Create
Active — Incoming calls
contacts
sessions
— Missed calls
>Multiple nos.
>Buddy or not >Supports re-dialing from log
Media
>Dial direct >Sort by name, number, time,
from contact
Voicemail
list
duration

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.

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• 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.
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

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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-up
o 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)

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• 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.

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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.

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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.

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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.

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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.

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2.7 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.

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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.

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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).

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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.

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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 devices


o Map a Primary phone to one or more Secondary phones
o 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 use
o blinking -> incoming call
o 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.

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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:

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• 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 calendar
o Time of day, day of week and calendar based call coverage is now available for
all extensions in addition to the coverage functionality in 7.0
o More flexibility in defining coverage for all the users
o 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.

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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.

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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.

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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.

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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.

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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.

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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

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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?

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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?

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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.

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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.

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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).

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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)?

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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.

Admin phone User phone Number Of


Feature Code mappable mappable Parameters
HANDS FREE TOGGLE 100 X X 0
MUTE BUTTON 101
VOLUME UP BUTTON 102

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Admin phone User phone Number Of
Feature Code mappable mappable Parameters
VOLUME DOWN BUTTON 103
SPEAKER TOGGLE 104
SOFT1 BUTTON 105
SOFT2 BUTTON 106
SOFT3 BUTTON 107
LCD UP BUTTON 108
LCD DOWN BUTTON 109
Feature 110 X X 0
Release 111 X X 0
OK 120 X X 0
ALPHA NUMERIC
TOGGLE 122
LCD LEFT BUTTON 154
LCD RIGHT BUTTON 155
DISPLAY IP ADDRESS 320
DISPLAY GATEWAY IP
ADDRESS 321
DISPLAY SUBNET MASK 322
DISPLAY NCP IP
ADDRESS 323
DISPLAY NCP2 IP
ADDRESS 324
DISPLAY CURRENT PBX
IP 325
DISPLAY SIP GATEWAY
INFO 326
DISPLAY SUBJECT INFO 327
DTMF VOLUME 329
CONFIG BEEP 330
SEND BEEP (beep feature
is a page sent between
phones – your phone tells
you who beeped you) 331
TOGGLE WARNING
TONE(busy tone generated
when phone is off hook past
a certain time – like a pstn
phone) 332
RESET PHONE TO
DEFAULTS 333
FORCE REGISTER 335
SOFT RESET 336
HARD RESET 337
DISPLAY DEBUG INFO 338
TOGGLE ASSERT
BEHAVIOR 339
USE 20MS FRAME SIZE 340
USE COMPRESSED
CODEC 341
USE 729B 342

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Admin phone User phone Number Of
Feature Code mappable mappable Parameters
USE SUBNET OPTION 343
DIAL BY URL 344
HELP 345 X X 0
REDIAL BUTTON 401
HOLD BUTTON 402
PROGRAM BUTTON 410
TRANSFER BUTTON 420
CONFERENCE BUTTON 430
CLASS OF SERVICE X X 0
OVERRIDE 433
USER PASSWORD (set
locally on phone only) 434
USER 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 837
VIEW 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
X 1 (bridged
Bridged Appearance 303 number)
Headset Enable/Disable 112 X X 0

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

Feature Code Admin User Number Of Feature Request


phone phone Parameters String
mappable mappable
Malicious Call Trace 119 0 *fc
Silent Monitor 425 X 1 *fc*monitor_num
Barge In 428 1 *fc*barge_in_num
FwdMail Toggle 440 X X 0 *fc
Transfer To Voicemail 441 X X 1 *fc*destination
(Coverage)
Park 444 X X 1 *fc*park_num
Do Not Disturb 446 X X 0 *fc
Directed Pickup 455 X X 2 *fc*pwd*pickup_num
Config Forward Universal 465 X X 0 in b-map *fc*fwd_number
1
Config Forward Busy 467 X X 0 in b-map *fc*fwd_number
1
Config Forward Ring No 466 X X 0 in b-map *fc*fwd_number
Answer 1
Config RemoteFwdUniversal 468 X X 2 *fc*src_num*fwd_num
CampOn 469 X X 1 *fc*destination Number
Attendant Serial Call 471 X X 1 *fc*transfer Destination

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Feature Code Admin User Number Of Feature Request
phone phone Parameters String
mappable mappable
Retrieve Voice Mail 600 1 *fc*user (optional)

(*user could be another


mailbox accessible to
this phone)
Personal Speed Dial 601 X X 1 (invoke) *fc*speed dial
2 (config) index*number
(*number is optional
and used to config the
speed dial through TUI)
System Speed Dial 700 X X 1 *fc*speed dial index
Block Caller Id Toggle 889 X 0 *fc
Block/Unblock CallId for 890 1 *fc*destination number
Current Call
Hunt GroupToggle* 971 X 1 *fc*hunt_group_num
Block Monitor/Barge 429 X X 1 *fc*destination
Call 125 X X 1 *fc*destination
Busy Lamp Field 126 X 1 *fc*destination
Config Button Map 127 2 minimum *fc*buttonNumber
*featureRequestString

Success codes for features are identified in the following table.

Code Description
1000: Success General success case used in many features
1001: Park Success: XXX Specific park success where the park number is returned
as well (XXX)
1002: Do not disturb enabled Do not disturb feature enabled
1003: Do not disturb disabled Do not disturb feature disabled
1004: 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 Description
9999: Unauthorized User is unauthorized to perform the feature
9998: No call for feature Mid call feature invoked outside of a call
9997: Invalid within call New call feature invoked within a call
9996: 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)

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Code Description
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 Error on callP during a request to connect to voice mail, where
mail 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.

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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 name
o Press the Clear button to sort by last name
o 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.

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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:

• Linear
o Single pass through list of members, bounce to coverage at end of list
• Circular
o 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 No Members Logged Out Members Busy Total


Type Members Timeout
Linear Go to Call Go to Call Coverage Go to Call Coverage Go to Call
Coverage Coverage
Circular Go to Call Wait for somebody logs Wait for somebody to Go to Call
Coverage in till Time out get free till Time out Coverage
Calling Go to Call Go to Call Coverage Queue this call and Go to Call

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Coverage wait for somebody to Coverage
get free till Time out

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

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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.

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• 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.

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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.

Ring...
Ring...
User1

Ring... Ring...
User 4 Ring...
User 2

To coverage
Ring... User 3

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.

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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.

To coverage
Circular Hunt Group

Ring...
User1

Ring...
Ring... User 2
User 4

Ring... User 3

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.

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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.
Linear Hunt Group

Ring...
User1

Ring... User 2

Ring... User 3

To coverage
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:

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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.

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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.

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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.

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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)

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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.

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• 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.

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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.

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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.

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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.

IP Messaging

SMTP, POP3, IMAP4


Lotus Notes

Domino
Email Server

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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.

Voice & Fax Messages


MWI synchronized in your email client
after each poll interval
MWI

Email Text POP3, IMAP4


Email Client
Optional
Text To Speech SMTP
Server
SMTP, POP3, IMAP4
Domino
Email Server

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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.

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Central Server
Local Office IM Local Office
(Regional) (Regional)

IM IM

Local Office Local Office


(Regional) (Regional)
IM IM

Local Office Local Office


(Branch) (Branch)

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 Directory


o 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 Sending
o 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

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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.
SIP
IP Messaging Server
Auto-delivery of
voice/fax messages
(SMTP)
3Com
IP Messaging
Email message
send, reply, forward
(SMTP) Call Builder

Email message or
MWI synchronization
(POP3/IMAP4)
Email Client Email Server

Speech Text

Windows XP or Windows Server

Scansoft Text Scansoft Text 3Com


SpeechWorks SpeechWorks IP Messaging
Speechify Speechify
Speech Speech
Voice Engine TTS Server TTS Adapter

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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.

SMTP,
POP3, IMAP4
Domino Network
Email Server IP Printer
Messaging
VCX Fax Machine
Email IP Telephony
Client
Analog FXS
IP Network Gateway
Digital
Gateway

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) √ √
French (Canadian) √ √
Italian √ √

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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?

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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.

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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 Report to Report to Report to


Scheduled? Screen? File? E-Mail?
All Subscribers No Yes Yes No
Subscribers by NPA No Yes Yes No
Subscribers by NPA-NXX No Yes Yes No
Subscribers by Company/Division No Yes Yes No
All Classes Of Service No Yes Yes No
ID Class Of Service No Yes Yes No
Administrators No Yes Yes No
Message Time Statistics No Yes Yes No
Port Usage Statistics Yes Yes Yes Yes
Mailbox Usage Statistics – Yes Yes Yes Yes
Regular Mailbox
Mailbox Usage Statistics – Auto Yes Yes Yes Yes
Attendant
Disk Usage by Message Type Yes Yes Yes Yes
Idle Mailboxes Yes Yes Yes Yes

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Report Name Report can be Report to Report to Report to
Scheduled? Screen? File? E-Mail?
Active Mailboxes Yes Yes Yes Yes
Call Processing by Mailbox Yes Yes Yes Yes
Call Processing by Port Yes Yes Yes Yes
Initialized Mailboxes Yes Yes Yes Yes
Failed 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?

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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.

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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.

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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.

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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.

office cell home


Incoming
call

Voice Mail
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?

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>.”

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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.

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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)

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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?

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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?

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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 recorded
o Standard message playback control options are available
• Press 2 to re-record
o Allows caller to re-record message
• Press 3 to mark message as urgent
o Message flagged as urgent
o Used to put in front of message queue and for notification procedures
• Press 4 to mark message as private
o Phone number of caller will not be available for mailbox owner to review
• Press 5 to enter call back number
o 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.

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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 recording
o The message recording is played back to the caller
• Re-record
o 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:

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• 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.

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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” Configuration
o Components installed on the single server:
 conference server/conference attendant
 presence server
 VCX database
 conferencing and presence database (master)
 web console server

• Dual Configuration
o 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 Configuration
o 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)

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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

Conference domain
Conferencing Conferencing Conferencing Conferencing

-Master DB
-VCX DB
-Web Console -Conferencing
-Slave DB
-Presence - Routing
-Routing

Primary Secondary

3Com VCX

IP Phone Convergence Center Client

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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.

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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 name


o 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 time


o 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 minutes
o 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 type
o 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

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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 interval


o 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 type
o Same as above except this is done when the participant leaves
• End of conference first warning
o 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 Indication
o Users receive a MWI if they fail to join an emergency conference when invited.
MWI is not available with instant conferences
o 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

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o Only administrators have the authority to configure emergency conferences
o Any user can configure an instant conference
• Source Address
o You can use any valid user address as the source address (“From” address) for
emergency conferences
o 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.

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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.

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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?

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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
firstname_lastname@3com.com”

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

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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.

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• *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

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2.8 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.)

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2.9 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.

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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.

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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.

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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.

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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 administration
o The manager role allows access to menu options that manage users and phone
extensions
o 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.

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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.

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2.10Reporting
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.

CDR
Reporting
Application
SIP Call
Processor

Regional Office IPT


(Primary)

SIP Call Accounting


Processor Super-CDR Server
ASN Encoded
XML Files

Merge Rules
Accounting ASN Encoded
Server XML Files
Call Record
(CDR) Server
Regional Office IPT
(Secondary)
Connection Rules Regional Offices
ASN Encoded
for pulling XML files
XML Files Secure FTP
from each Accounting Server Branch Offices

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 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

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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:

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o Average call duration for hunt group
o Call Distribution for hunt group by agent by day
o Call Distribution for hunt group by agent by hour
o Call Distribution for hunt group by day
o Call Distribution for hunt group by hour
• Monitor Barge-In

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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 grid
o Specify which call data fields are to be included in the call data grid
o 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 grid
o 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.

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An example of IP Messaging data in the Data Grid is shown below.

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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.

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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.

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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.

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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.

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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.

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An example of a Malicious Call Trace report is shown below.

An example of a Gateway Utilization report is shown below.

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An example of a Call Distribution for Hunt Group by Agent by Day report is shown below.

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The VCX Call Reporting application also supports graphical reports, as shown in the Calls per
Hour report shown below.

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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.

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2.11 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

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• Media Gateways
• Integration available with EMS for IP Convergence Applications:
o Using scheduled operations for VCX upgrades
o Using data collector to collect call statistics
o Using data collector to collect channel stats on gateways
o 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.

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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.

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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.

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Implementation
Fully describe the implementation process.

3Com adheres to a five (5) phase approach to Project Management, covering Sales, Planning,
Implementation, Testing and Servicing.

Sales Implement Support

PROJECT MANAGEMENT

Phase I Phase II Phase III Phase IV Phase V


Service and
Proposal Plan Deploy Review Design
Test
Monitoring

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

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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.

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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.

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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.

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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;

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• 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 framework
1999 – Softswitch deployed at AT&T
2001 – Industry’s first SIP-based Softswitch
2001 – SIP Softswitch at MCI
2003 – Introduction of VCX IP-PBX for multi-site medium to large enterprises
2003 – First with SIP based convergence applications suite
2004 – 16,000+ NBX systems shipped
2004 – VCX Release 5 available
2005 – VCX Release 6 available
2006 – 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.

GK DIR/PROV AS/BSS
3Com
Softswitch

TC1000 TC1000

LEC IP
Tandem Backbone LEC
Switch
Calling Called
Party Party
Tandem
Tandem Switch
Switch
Tandem
Network
Local Network Local Network
Transport Network

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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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4 Pricing

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