VoiceCon Spring 2007 - Conference Presentation

VoiceCon Spring 2007 - Conference Presentation
An NEC Solution for
VoiceCon 2007
Request for Proposal
For an IP Telephony System
Electronic Copy
Submitted December 1, 2006
Submitted By:
Chuck Ferguson
Assistant General Manager
NEC Unified Solutions, Inc.
214-262-6026
[email protected]
VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
NEC Unified Solutions Inc., a leader in integrated communications solutions for the
enterprise, delivers the industry’s most innovative suite of products, applications and
services that help customers achieve their business goals. With more than a century of
communications and networking expertise, NEC Unified Solutions, Inc., a subsidiary of
NEC America and affiliate of NEC Corporation (NASDAQ: NIPNY), offers the broadest
range of communications services and solution choices, flexible product platforms and
applications, and an open migration path to protect investments. NEC Unified Solutions,
Inc. serves the Fortune 1000 and customers across the globe in vertical markets such as
hospitality, education, government and healthcare. NEC Unified Solutions is a Cisco
Systems Gold Certified Partner, and Advanced Technology Partner and an IP Telephony
Specialization Partner.
NEC Unified Solutions, Inc., (“NEC”), is grateful for the opportunity to provide a proposal
and response to your organization for an IP Telephony System. While NEC realizes
that, under certain circumstances, the information contained within our response may be
subject to disclosure, NEC respectfully requests that all pricing, engineering design, and
unique or specific hardware configurations provided herein be considered proprietary
and confidential, and as such, not be released for public review. Please notify NEC
promptly upon your organization’s intent to do otherwise.
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VoiceCon Spring 2007
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Table of Contents
Tab 1: System Performance Requirements
Tab 2: System Pricing
Tab 3: Appendices
Detailed Features Listing
Voice Terminals PowerPoint
System Diagrams PowerPoint
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Request for Proposal
for an IP Telephony System
PART 1: System Performance Requirements
Submit Part 1 responses in MS Office WORD file format except
when otherwise noted.
1.0.0
System Overview
The VoiceCon Company plans to install a new IP Telephony System (IPTS)
network to support its newly constructed Headquarters (HQ) facility, a
Regional Office (RO), and three Satellite Branches (SBs) with Survivable
Remote Gateway (SRG) capabilities.
Dedicated local IPTS call telephony servers must be installed at the HQ and
RO facilities. All proposed call telephony servers must independently support
all generic software features for the proposed IPTS model(s) as required in
Section 5 of this RFP. The three SBs will be configured as survivable
remotes behind the HQ IPTS call server with local trunk circuit services
(Note: Survivability requirements for the SB facilities are identified later in
this section). The proposed IPTS network solution may include a single fully
distributed IPTS or no more than two IPTSs (each housed at HQ and RO
facilities). If a single IPTS is proposed the distributed call servers must
function and operate independently of each other, and support all generic
software features as required in Section 5 of this RFP.
The HQ IPTS call server will initially support 1,360 station users at the HQ
and three SB facilities. The RO IPTS call server will initially support 250
station users. See Figure 1 for an overview of the VoiceCon IPTS network.
See Figures 2 – 6 for port capacity requirements at each of the five VoiceCon
facilities.
VoiceCon anticipates 50% station user growth at the HQ and RO facilities,
only, and the proposed IPTS network solution must accommodate this
growth without replacement of any installed hardware/software.
There is no anticipated growth at the SB facilities. A centralized messaging
system will be housed at the HQ facility and must be capable of supporting
station users located at all VoiceCon facilities (HQ, RO, and SBs).
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Request for Proposal
for an IP Telephony System
Figure 1
Voicecon IPTS Network
HQ IPTS
1200 stations
SB2 SRG
50 Stations
SB1 SRG
100 Stations
WAN
RO IPTS
SB3 SRG
10 stations
HQ: Headquarters
RO: Regional Office
SB: Satellite Branch
IPTS: IP Telephony System
SRG: Survivable Remote Gateway
250 Stations
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Request for Proposal
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Figure 2
HQ Port Requirements
HQ IPTS
1200 stations
6 Local T1 circuits
7 Long Distance T1 circuits
5 PFTS circuits
25 Emergency Analog GS/LS Circuits
Figure 3
RO Port Requirements
RO IPTS
250 stations
2 Local T1 circuits
2 Long Distance T1 circuits
2 PFTS circuits
10 Emergency Analog GS/LS Circuits
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Figure 4
SB1 Port Requirements
SB1 SRG
100 stations
1 Local T1 circuit
2 PFTS circuits
5 Emergency Analog Circuits
Figure 5
SB2 Port Requirements
SB2 SRG
50 stations
10 Analog LS/GS circuits
2 PFTS circuits
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Figure 6
SB2 Port Requirements
SB3 SRG
10 stations
5 Analog LS/GS circuits
1 PFTS circuit
VoiceCon has plans to install at all of facilities LAN/WAN cabling and a
transport infrastructure that will fully satisfy the stringent requirements of IP
Telephony communications for all intra-premises and inter-premises call
control and voice communications transmissions. Each location will be
equipped, at minimum, with a 1-Gbps Ethernet backbone. The local wiring
closets will house 10/100/1000 Mbps Ethernet switches equipped with Power
over Ethernet (PoE). Multi-service routers will be installed at all locations to
support a MPLS WAN installation. All Ethernet switches and IP WAN routers
will be equipped and programmed to satisfy QoS and security standards
necessary to support voice communications acceptable to VoiceCon.
Pertinent bandwidth, latency, packet loss, and echo issues will be addressed
in the design and implementation.
Each station user’s work area will be supported by four (4) four-pair,
Category 5E cable wiring with one (1) RJ-11 wall connector and three (3) RJ45 wall connectors to the local wiring closet. The RJ-11 and RJ-45
connectors will be either wall mounted or mounted in the modular furniture
throughout the office environment. VoiceCon plans to run its IP Telephony
system over this cable infrastructure. NOTE: The proposed IP Telephony
system must be able to support a limited number of non-IP stations, e.g.,
analog telephones, requiring a RJ-11 connector. The proposed system can
use either circuit switched port carriers or media gateways to support analog
communications terminal equipment.
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Vendor Response Requirement
Based on the RFP requirements in this document prepare a simple network
diagram that illustrates the proposed IPTS network design. Include in the
diagram the brand name/model of the IPTSs, all circuit switched port
carrier/media gateway equipment, the brand/name of the HQ-located
systems management and messaging system. The diagram must be
prepared and submitted in a separate file using MS PowerPoint
format (identify the file as part of your electronic proposal
submission). In addition copy/paste diagram in the submitted MS
WORD file proposal here.
NEC Response: NEC Proposes the Univerge SV7000 Architecture to meet the
requirements of this RFP. The SV7000 is configured with a node at the Headquarters
and a second node at the Remote Office interconnected by NEC’s Fusion Call Control
Signaling (FCCS) Protocol over the IP WAN. The FCCS protocol provides for a onesystem look and feel for all of the VoiceCon Users. A user will be able to visit any node
and log-in to an available IP telephone. That telephone will immediately pick up all of the
features and button appearances assigned to the user’s primary telephone. Any
incoming calls will immediately be routed to this new location.
The AD-120 Voice Mail System located at the Headquarters will support all users at all
locations for call answering, auto attendant, and voice mail retrieval. Additionally, the
AD-120 is configured with Unified Messaging so that all voice mail messages are
available through the Microsoft Outlook email interface.
The Survivable Remote Media Gateway Controllers (SRMGC) configured for the three
Satellite Branches will support all of the features and functionality of the HQ node
SV7000 if the WAN connection between them fails. It is even possible to use the Public
Switched Telephone Network (PSTN) to communicate between the Satellite Branches,
the Headquarters and the Remote Office is the WAN should fail.
The system will be managed from the MA4000 located at the Headquarters. This serverbased management tool is configured to handle all move, add and change activity for all
five locations. Additional modules are available to handle the traffic reporting, call
accounting and work order- trouble ticket tracking for the system.
NEC’s Proposed IPTS Network Diagram
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NEC Univerge
SV7000 S
SV7000 T
Redundant
MA4000
Management
System
AD-120
Voice Messaging/
Unified Messaging
System
MPS Gateway
Analog Stations and Trunks
NEC Media
Gateway (PRI)
HQ LAN
Survivable Remote Media Gateway Controller
(SRMGC)
Survivable Remote Media Gateway Controller
(SRMGC)
SB1 LAN
SB3 LAN
WAN
Survivable Remote Media Gateway Controller
(SRMGC)
NEC Univerge
SV7000 S
SV7000 T
Redundant
SB2 LAN
RO LAN
Please refer to the attached PowerPoint file for additional network detail diagrams.
1.0.1
LAN/WAN Requirements
VoiceCon has not yet decided on the make/manufacturer of its new LAN/WAN
communications equipment.
Vendor Response Requirement
Indicate if the proposed IPTS solution for the HQ and the remote facilities
requires manufacturer-specific LAN/WAN communications equipment to
support any or all of the following voice communications operations or
functions: call processing, switching, routing, PoE, media gateway, QoS and
security. If responding in the affirmative, only, identify the make and model
of the necessary switch/router equipment and the reason for its requirement.
NEC Response: The proposed solution consisting of NEC Univerge SV7000
equipment does not require any specific manufacturer’s LAN or WAN
equipment, except that QoS must be available for the VoIP to function
correctly.
1.1.0 Basic IPTS Requirements
The proposed IPTS equipment should be in current production and operating
as part of a commercial system for at least five (5) customers in the USA.
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Vendor Response Requirement
State if the proposed IPTS equipment satisfies this commercial availability
requirement. If the IPTS model has not yet been shipped and installed in a
commercial installation, state expected availability date. Also provide an
estimate of the number of IPTS solutions (same model as proposed)
currently installed and operating in the USA.
NOTE: All proposed system hardware and software must be formally
announced as of VoiceCon Spring 2007 to be accepted by VoiceCon in
response to this RFP. This is a mandatory requirement to submit a
RFP response.
NEC Response: The Univerge SV7000 is currently available for delivery and is currently
installed in over 100 locations in the US and many more Worldwide. All proposed
hardware and software will be formally announced as of VoiceCon Spring 2007.
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1.1.1 Single System Image
The proposed IPTS network should provide a Single System Image across
VoiceCon HQ, RO and SB facilities. The Single System Image should include,
but not be limited to, the following:
1) 5-digit dialing between all station users;
2) High degree of transparent operation across all VoiceCon facilities for
station, attendant, and system features (see RFP Section 5: Call Processing
Features);
3) HQ-located centralized systems management solution using a single
unified database for all station user profiles, equipped system design, and
system-level operations;
4) Network-wide attendant operator services across all VoiceCon facilities,
including the ability to support a centrally located attendant pool;
5) Shared messaging system resources;
6) Automatic alternative routing across the network for all voice calls
(station-to-station and PSTN trunk connections).
Vendor Response Requirement:
Provide specific answers to each of the following questions:
1. Is the proposed IPTS network solution a single system solution or multiple
systems intelligently networked?
2. Does the proposed IPTS network solution fully satisfy all six (6) of the
stated Single System Image requirements? If not, explain which of the
requirements are not satisfied?
NEC Response: The proposed Univerge SV7000 is a single solution
across the HQ, SB1, SB2 and SB3 sites with an intelligent network
connecting the RO. The network is NEC’s Fusion Call Control Signaling
(FCCS) Network and this protocol gives the appearance of a single
system solution for all users. All programming for all locations is
performed at the HQ site.
Yes, the proposed Univerge SV7000 meets all 6 of the above criteria for a single system
image.
1.1.2 Enhanced 911 (E911) Services Support
It is mandatory that the proposed and installed communications system
support E911 services provided by a public safety answering point (PSAP) as
defined by FCC regulations. All VoiceCon IPTS network locations addressed
by this RFP are served by the same PSAP.
All VoiceCon IPTS station user E911 calls must be directed to their local PSAP
for call handling and response regardless of location, i.e., facilities remote
from the primary call telephony server. If more than one E911 solution is
available for the proposed IPTS network configuration clearly specify the
solution that is included in the price proposal.
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Vendor Response Requirement:
Confirm that the proposed communications system solution supports E911
service for all user stations (IP and analog) at each of the VoiceCon facilities.
In the response briefly explain how E911 service requirements are
supported, specifically addressing each of the following questions:
1) A description of any optional hardware/software equipment included in
the pricing proposal, and if a peripheral server is required who is
responsible for its purchase?
2) How are station user moves/adds/changes reported to the E911
provider?
3) What degree of specificity station user location is identified to the E911
PSAP? Desktop work area, local switch room, work floor, other?
NEC Response: The proposed SV7000 supports E911 service at all locations. The ANI
feature for Enhanced 911 (E911) Outgoing Connections is available to the following
connections:
•
Call from a Station by Least Cost Routing (LCR) or MF Signaling.
•
Call from an Attendant Console by LCR or MF Signaling.
The SV7000 supports gateway PRI and analog CO trunks for E911. Locations with the
SR-MGC are provided with PFT to local trunks in case of IP network disruption.
1) No optional hardware or software equipment is required to support E911.
2) All station moves/adds/changes are recorded in the MA4000 and incorporated
into the NENA compatible database for transmission to the local PSAP. The file
can be updated by printing and delivering it, by copying the file to CD or Floppy
Disk or by FTP. The actual transfer of the information to the PSAP depends on
the arrangements available from the PSAP.
3) Each telephone location can be specified to the level required to accurately
reflect the telephone’s location.
1.1.2.1 E911 and Station Moves
It is required that station user moves behind the proposed IPTS solution be
tracked dynamically in real time for E911 services support.
Vendor Response Requirement:
Indicate if the proposed E911 solution satisfies this desirable capability and
indicate how often the database updated. If an alternative E911 solution is
available that satisfies this capability, but is not included as part of the
overall IPTS solution and pricing proposal, briefly describe this option and the
incremental costs to purchase and install beyond the proposed solution.
NEC Response: No, station moves are not reported automatically to the E911 PSAP.
This is a potential capability of the MA4000 System but the PSAP requirements are
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unique to each PSAP and so the information must be delivered manually as described in
the response to the previous question.
1.2.0
Proposed Communications System Design
The proposed communications system may only be based on either of the
two following architecture technology designs:
•
•
Single system design based on true peer-to-peer distributed call
processing topology, i.e., identical or similar call telephony servers
located at all VoiceCon facilities (HQ, RO, SBs)
Intelligently networked multiple system design based on identical or
similar call telephony servers located at VoiceCon HQ and RO facilities,
and survivable remote gateways at VoiceCon SB facilities configurable
behind the HQ call telephony server.
Only a supplier’s most current generation hardware/software solution will be
acceptable. No refurbished equipment is acceptable.
NOTE: There is no preference for either the single or multiple system design
if all 1.1.1 Single System Image requirements are satisfied.
Vendor Response Requirement:
Briefly describe your proposed solution, referring to the diagram from RFP
Clause 1.0.0 when applicable.
Limit your response in this section to the following high level information as
details are requested in following sections:
1. Product and model name(s) for the IPTS(s) and messaging system.
2. Identify proposed solution as a single system or multiple system
design.
3. For each network location specify the product/model used to
support station/trunk call processing and switching operations under
normal operating conditions.
4. Identity the software release for each product/model proposed
5. Provide the product/model introduction dates.
NEC Response: The NEC Univerge SV7000 system proposed is an intelligently
networked multiple system design consisting of two SV7000 nodes (one at HQ and one
at the RO) interconnected using NEC’s Fusion Call Control Signaling (FCCS) protocol
over the IP WAN. The Satellite Branches are configured with Survivable Remote Media
Gateway Controllers (SRMGC) which are controlled and configured from the HQ
SV7000.
Under normal operating conditions, the SV7000 located at the HQ controls all call
handling functions for the HQ, SB1, SB2 and SB3 locations. The SV7000 located at the
Remote Office controls all call handling functions for the Remote Office.
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The SV7000 and the SRMGC were introduced in 2004 and the proposed configuration
will use Software Version 20.5 introduced in November 2006, unless the release of
Version 21 occurs before system delivery. In that case, version 21 will be provided along
with a full description of all feature additions and enhancements at no additional charge.
1.3.0 System Design Platform
The proposed system solution may be based on either of the following two
architecture system design:
•
•
Converged TDM/IP: call telephony server supporting LAN/WAN
distributed circuit switched port interface cabinets with equipped
media gateway interfaces for IP port connectivity
Client/server: call telephony server supporting media gateway
equipment (server-embedded, standalone, switch/router-equipped or
desktop) for non-IP port connectivity
Vendor Response Requirement:
Briefly and clearly describe the architecture and design elements of the
proposed IPTS solution. Include in your basic system description information
about the following common equipment hardware elements:
1. Type of architecture design (converged or client/server)
2. Call telephony server and associated common control equipment
3. If applicable, circuit switched port interface equipment housing TDM
port interface circuit cards and media gateway boards.
4. If applicable, LAN-connected media gateways (server-embedded,
standalone, switch/router-equipped, desktop
NEC Response: The NEC Univerge SV7000 is a client/server architecture design using
the SV7000 S and T modules for call control and system supervision. These modules
are 1U rack mount modules and are configured as special-purpose servers (cannot be
reconfigured to support other server-based applications). The advantage of using a
special-purpose server is that all of the overhead required to keep track of multiple
applications running concurrently can be removed allowing the server to concentrate its
time on call processing functions.
The Univerge SV7000 uses LAN connected stand-alone Gateways for interconnecting to
analog stations and trunks and to ISDN PRI trunks. There are additional gateways for
conferencing. To reduce costs and provide for a more compact installation, NEC has
configured all of the gateways for mounting in 1U rack mount shelves (two modules per
shelf). However, there are also desk-top gateways available for applications such as fax
machines located at a distance from the IDF or equipment closet.
1.3.1 Common Control
The primary common control complex of the proposed IPTS should be based
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on a standalone call telephony server or a call processor blade that is
embedded in common equipment that functions as a call telephony server.
The physical equipment may either be a fully bundled proprietary
hardware/software offering that is factory configured or third party
equipment provided by VoiceCon that is capable of running proposed
proprietary call processing software without any service degradation.
Any and all of the proposed primary common control call processor elements
used to provide call processing functions must be proposed in a redundant
duplicated design with seamless switchover operation between active and
standby control elements, i.e., all active call connections must remain up
during switchover in case of failure or major alarm states and new calls setup without delay. The secondary standby control element must be local
to the primary, i.e., physically located at the same VoiceCon facility.
The secondary standby cannot be located at a remote VoiceCon
facility. This takes into consideration the possibility of simultaneous
primary control and LAN/WAN link failure that affects
telecommunications services to station subscribers. This redundancy
requirement does not apply to local survivable processors at SB
facilities where primary control is located at the HQ facility.
Solutions based on fully dispersed call processing system designs,
i.e., primary control elements at HQ, RO, and SB facilities, however,
must conform to the local redundancy requirement wherever a
primary control element is installed.
The overall common control design may be based on a load sharing design in
which any call telephony server/processor blade may be programmed to
function in primary and secondary backup modes. All primary common
control elements must be capable of supporting required equipped and wired
capacities at time of installation.
The call processing rating for the proposed primary and secondary
IPTS call server(s) or equivalent(s) must minimally support the
following call processing ratings at each of the following VoiceCon
facilities:
HQ: 35,000 Busy Hour Call Completions (BHCCs)
RO: 10,000 BHCCs
SB1: 5,000 BHCCs
SB2: 2,500 BHCCs
SB3: 1,000 BHCCs
Vendor Response Requirement:
Provide a brief description of the common control design you are proposing in
terms of design platform: call telephony server, call processor blade
(including necessary housing), third party server. Confirm that the
duplicated common control requirement is fully satisfied by the proposed
solution, and identify any feature/function that is not available if a standby
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(back-up) call processing element must be activated in case of a primary
element failure.
NEC Response: The NEC Univerge SV7000 is a special purpose telephony server
equipped with full local redundancy. To provide redundancy the SV7000 T module is
connected to a second T module via a front-panel flat cable. In case the primary T
module fails, the secondary module has been updated continuously and can
immediately take control of the system with no loss of calls in progress. The SV7000 S
module and the redundant S module are connected to the LAN and load share the call
processing functions they perform. Thus if one S Module fails, the other(s) can take over
the functionality. Up to five S Modules may be configured but only two are required in the
proposed configuration.
If a failover occurs in either the S or T modules, all calls currently in progress continue
uninterrupted. All features and functions continue to be available and all programming of
station configurations is maintained. The only loss is for calls where the call setup has
not completed. In those cases, the caller receives reorder tone and must dial again.
1.3.2 CPU Make/Model
Vendor Response Requirement:
Identify the make/model of all proposed common control CPU(s) and
associated BHCC rating for the configured system.
NEC Response: The processor used in both the S and T modules is an Intel Pentium M
2.2 GHz processor. The basic processor architecture is centralized. The Maximum Busy
Hour Call (BHC) attempt rating is 46,000 BHCA.
1.3.3 Call Processing O/S
Vendor Response Requirement:
Identify the primary operating system of the common control call processor.
A version of Linux is preferred, but not mandatory.
NEC Response: The SV7000-T operating system is NEC proprietary.
The SV7000-S contains a Linux-based operating system.
1.3.4 Memory
Vendor Response Requirement:
Briefly describe the main memory design and storage elements and capacity
for both the generic software and customer database as proposed.
NEC Response: The NEC Univerge SV7000 S and T modules each have 512 MB RAM
memory. This memory is used for both generic and database on-line memory. Each
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module also has a Flash Memory card of 512 MB that backs up the RAM and restores
the memory contents if the system resets because of power failure.
1.3.4.1 Database Integrity
Vendor Response Requirement:
How does the proposed IPTS solution maintain the integrity of the customer
database between back-ups?
NEC Response: The Flash Memory Card inserted into the S and T modules are used to
maintain the Generic and database memory of the system between back-ups.
1.3.4.2 Database Information Loss
Vendor Response Requirement:
Identify under what circumstances can customer database information
(configuration, messages, logs, etc.) be lost during back-ups
NEC Response: The Customer information can be lost if a change is made and then the
command to “write to flash” is not given. In this case, the flash memory will not know of
the change and so when it is backed-up, the change will not be included.
1.3.4.3 Database Backup Scheduling
Vendor Response Requirement:
How often should the customer database be backed up? Specify if it is a full
or incremental backup and the time the process takes.
NEC Response: The system database should be “written to flash” after every change is
made. This incremental process does not interfere with system operation and takes only
a few minutes (including on-line and redundant flash cards). Backing up the flash
memory is an off-line process where the flash card is removed from the module and
inserted into a PC. The information from the flash can then be backed-up to any PC or
server on the network. NEC recommends that full back-ups be done on a periodic basis
that fits with the overall back-up strategy of the LAN/WAN infrastructure.
1.3.4.4 Data Purging/Archiving
Vendor Response Requirement:
Describe the mechanism for data purging and archival, including storage and
retrieval of archived data.
NEC Response: Data purging and archival is a network function. Once the Flash
memory is backed-up to the network, data purging and archival should be done in
accordance with the local policies and procedures.
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1.3.5 Power Supply
Vendor Response Requirement:
Briefly describe common control power requirements and the integrated
power distribution design. Indicate if the power supply is dependent on either
an AC or DC current source.
NEC Response: The following describes common control and port cabinet/carrier power
requirements.
Main Power Source
The UNIVERGE SV7000 requires an operating power of 100-120V AC. For
greater system reliability, it is recommended that connection of all the AC power
cords from SV7000 and PIRs to one AC tap, for securing protective common
grounding.
Note: According to the system configuration, connect the AC power cords to a
power supply that can supply enough power for the SV7000 and PIRs.
Additionally, power backup is recommended so that telephony will not be lost if a
momentary power fluctuation occurs.
Install Type1 grounding (below 10 ohms) for the AC power taps.
AC taps or power supply unit connected to power distribution board must be
provided by the user.
Power Consumption
The UNIVERGE SV7000 System operates on 100-240V AC. AC-to-DC
converters are mounted in each UNIVERGE SV7000 9U PIR. Upon receiving the
–48V DC power input from the AC power packages, the DC-to-DC power
converters, also mounted in the PIR, converts it to the various DC voltages
needed and in turn supplies those voltages to the circuits cards mounted within
the PIR.
Note: The UNIVERGE SV7000 9U PIR circuit cards are actually rated at -48V
DC, -5V DC, +80V DC.
Note: The PIR is a Port Interface Rack which can be configured on the SV7000
to support line and trunk cards in a hybrid TDM/IP environment. There are no
PIRs proposed for VoiceCon.
1.3.5.1 Power Safeguards
Vendor Response Requirement:
Describe any power failure safeguards that are included in the IPTS design.
Briefly describe what happens to system operation during a power failure
NEC Response: If power is lost to the NEC Univerge SV7000, all processing stops in all
modules that lose power. The exception is that there are PFT configurations of those
gateways that include both analog stations and analog trunks. In those instances, the
analog trunks are connected to analog stations upon power failure.
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1.3.5.2 Power Backup
Vendor Response Requirement:
Is the proposed IPTS solution equipped with standard UPS hardware, and if
so how long can the system run on it? If not, what UPS requirements are
recommended?
NEC Response: The NEC Univerge SV7000 does not include a UPS. NEC can
configure a UPS system for the proposed SV7000 but has not in this case. NEC
recommends that a UPS system with a minimum 20 minute back-up battery be used so
that momentary power fluctuations will not impact service. Additionally, NEC
recommends that the entire SV7000 system including the S and T modules, all gateways
and all LAN switches with POE for telephones be connected to local UPSs and a
generator for long term power protection.
1.3.6 Ethernet Call Control Signaling Links
Vendor Response Requirement:
Identify for each active and standby call telephony server the number of
available and configured RJ-45 Ethernet LAN uplink interfaces for call control
signaling to LAN-connected cabinets/carriers and/or standalone ports.
Include a brief description of how the physical Ethernet connection is
provided: dedicated circuit board; daughterboard; fully integrated RJ-45
connector, et al.
NEC Response: Each module and gateway in the Univerge SV7000 system has a
single RJ-45 connector built-in to provide connectivity to the LAN.
1.3.7 System Clocks
Vendor Response Requirement:
Identify the number and type of internal system clocks that are available and
configured.
NEC Response: The Univerge SV7000 uses a real time clock similar to those used in
Personal Computers. There is a single clock in each SV7000 T module.
1.3.8 Redundant system design elements
It is desirable to have a highly redundant system design, especially as it
relates to common control elements necessary for call processing,
maintenance, and administration operations.
Vendor Response Requirement:
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Specify the level or degree of redundancy included in your proposal for each
of the following listed common control elements. For example, full duplicated
back-up, standby load sharing, seamless switchover, cold standby, et al.
•
•
•
•
•
•
•
•
•
•
Primary call processor
Main system memory
Customer database memory
RJ-45 Ethernet uplinks to network
Power supply
Tone generators
Call classifiers
Registers
DTMF receivers
I/O interfaces
NEC Response: The SV7000T (Telephony server) redundancy provides a standby
SV7000T server. The SV7000S (SIP server) redundancy provides n+1 clustering of
SV7000S servers, up to 5 servers.
Redundant System Design Elements
Primary call processor
Main system memory
Customer database memory
RJ-45 Ethernet uplinks to
network
Power supply
Tone generators
Call classifiers
Registers
DTMF receivers
I/O ports
Full Duplication
Full Duplication
Full Duplication
One per Module
One per Module, Full
redundancy for PIR
Full Redundancy
Full Redundancy
N+1 Redundancy
N+1 redundancy
One per Module
1.4 Local Survivability
It is important to VoiceCon that station users at all network locations have
access to telephony services at all times. This includes 100% of generic
software features and trunk circuit access to a local exchange carrier. For
this reason it is highly desirable that station users at VoiceCon’s SB facilities
have access to telephony services in case of HQ-SB WAN link failure due to
switch, router, or private network transmission service issues, or HQ
common control failure for any reason.
It is preferable that standby telephony services be provided by an on-site call
processing option. A less desirable, but acceptable, emergency option is an
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alternative PSTN-based call control signaling link, but only if an on-site call
processing option is not available as part of the system solution. It is
also highly desirable that the standby call processing option provide stations
users with the same level of telephony services, i.e., station,
attendant, and system features, supported by the HQ IPTS at the medium
(50 stations) and large (100 stations) SB facilities. For the small (10
stations) SB facility it is acceptable that POTS-like survivability (dial tone,
PSTN trunk access, intercom calls, basic features such as Hold and Transfer)
is supported. Please note that Power Failure Transfer Station (PFTS) is not
acceptable as the local survivability option at the small SB facility.
SB facility local survivability for any disruption due to any
circumstance (common control failure and/or LAN/WAN incidents) of
HQ-based IPTS call control signaling is a mandatory requirement for
proposal submission.
Vendor Response Requirement:
Describe the proposed local survivability solution that satisfies the stated
requirements. Include a description of all proposed and priced local
survivability options (including any and all required hardware, software, and
PSTN transmission services necessary to implement the option) for each of
the three SB facilities: 10 stations, 50 stations, 100 stations.
NEC Response: The NEC Univerge Survivable Remote Media Gateway Controller
(SRMGC) is a single 1U module that is located at the remote location and is under full
control of the SV7000 T server during normal operations. When a failure occurs that
isolates the remote location from the main location, the SRMGC assumes control of all
IP telephones and Gateways at the remote site and provides full feature functionality
consistent with the types and number of gateways at the location.
The exception to this is the attendant console at SB1. Because the attendant console is
controlled by a server at the HQ, the attendant console will default to stand alone IP
telephone functionality. This can still be the main answering point for the incoming calls
on the ISDN PRI but all call handling will be done without the benefit of the console
functionality.
In the proposed configuration, each Satellite Branch (SB1, SB2 and SB3) has an
SRMGC which is controlled by the SV7000 at the Headquarters site. The Gateways
configured at each site are in accordance with the requirements of Table 5 in Section
2.3.
Based on table 5, SB1 will have a single ISDN-PRI Gateway plus 7 analog trunks; SB2
will have 12 analog trunks; and, SB3 will have 6 analog trunks.
1.4.1
Survivable IPTS Features/Services
Vendor Response Requirements:
Identify any required generic software feature (See Section 5.0 Call
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Processing Features) not available or operational when the local survivability
solution is activated at VoiceCon’s 50 station and 100 station SB facilities.
Also identify any type of station user equipment (instruments, consoles,
softphones, wireless communications devices, et al) not supported in standby
survivability mode at these two facilities.
NEC Response: All of the generic software features that are available under normal
conditions will be available when the system activates local survivability. All of the station
user telephones, wireless telephones and softphones will retain the same functionality
as normal. However, since the voice mail system is not directly accessible, it is possible
that calling to voicemail may be blocked by the limitation of available trunks.
The exception to this is the attendant console at SB1. Because the attendant console is
controlled by a server at the HQ, the attendant console will default to stand alone IP
telephone functionality. This can still be the main answering point for the incoming calls
on the ISDN PRI but all call handling will be done without the benefit of the console
functionality.
1.4.2
Local Survivability Failover and Switchback
Vendor Response Requirements:
For each of the SB facilities is failover to the local survivable call processing
option seamless, i.e. no interruption of in-process telephony services, for any
or all stations users if WAN connectivity is disrupted to the HQ IPTS? Indicate
in answer if there is delay for implementing new calls immediately after the
WAN disruption. Also describe the switchback process when HQ facility IPTS
call control is again available via the WAN, specifying if the process is
automatic or manual and how long the process takes to implement. Are
connected calls and voice operations at the remote facility affected in any
way by the switchback process and how soon can new calls be implemented?
NEC Response: When the SV7000 system fails over to the local survivability mode the
change will terminate every active call. This is because the telephones now need to
register with the SRMGC in order to get service. This registration process may take up to
two minutes at SB1, up to a minute at SB2 and just a few seconds at SB3. Once the
telephones have registered to the SRMGC their display will return to a normal mode and
the telephone may be used to place calls.
When the WAN again becomes available and a connection can be established between
the SV7000 T module and the SRMGC, the switchback process will begin automatically.
All currently idle telephones and gateways will be instructed to register with the SV7000.
Telephones and gateways involved in calls will be unaffected until the call disconnects.
As each telephone becomes available, it will automatically register with the SV7000. The
registration process takes just a few seconds for each device but there may be delays
depending on how many devices are trying to register simultaneously. As soon as the
telephone has registered, it may be used to make calls.
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Survivable Messaging Services
It is desirable that remote station users at the RO and SB facilities have
access to messaging services if there is a WAN link disruption to the HQ
messaging system.
Vendor Response Requirements
Does the proposed IPTS network and messaging solution satisfy this
requirement if WAN connectivity between HQ and any of the other facilities
(RO, SBs) is not available? Briefly describe how messaging services would be
implemented and accessed by remote station users in emergency situations.
The minimum messaging services function in survivable mode should include
voice mailbox access by station users.
NEC Response: When one of the Satellite Branches goes into Survivable Mode, it is
still possible to dial the voicemail server using either the ISDN PRI (at SB1) or an analog
trunk (at SB1, SB2, or SB3) to reach the HQ site.
1.4.4
Network Failover Resiliency
In the unlikely event the redundant common control complex (primary active
and secondary backup) at either HQ or RO facilities become nonfunctional
due to extreme system failure or catastrophic circumstances, e.g., fire,
VoiceCon requires implementation of a resilient network failover process.
This process requires that all local station users and media gateway
equipment configured behind the nonworking common control complex
automatically re-register to designated emergency call telephony server(s) at
either the local or remote facility for continuity of telephony services. For
this reason it is necessary that the designated emergency call telephony
server(s) located at the HQ/ RO facility be capable of supporting sufficient
port capacity requirements in the event of a failover.
Vendor Response Requirements
Does the proposed IPTS solution support network failover resiliency in case
of a catastrophic common control failure at either the HQ or RO facilities? If
affirmative, describe the failover process, optional hardware/software and/or
WAN transmission requirements to implement, and the time required for the
network failover to be implemented before telephony services are available.
Indicate if the proposed IPTS solution can support more than one network
failover design.
NEC Response: The NEC Univerge SV7000 currently allows only a primary and a
secondary registration for an IP device (gateway or telephone). In software expected to
release within 2007 it may be possible to extend that to up to 5 levels of registration.
Multiple registrations work similarly to failover from the SV7000 to an SRMGC. That is,
when the primary system fails, the telephone recognizes loss of registration (loss of
keep-alive signaling) and begins to register with the next server on its list. Successful
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registration takes only seconds but may be delayed by the volume of devices registering
simultaneously.
For this process to work, the network must be configured with multiple call servers
(SV7000 T and S Modules or SRMGCs). These multiple servers can be located at any
point on the LAN/WAN provided that the servers can pass packets back and forth with
QoS and minimum delay. Then each IP device (telephone and gateway) is configured
with the list of available servers. Note that the servers must each have a license for each
telephone that may connect to them simultaneously.
This proposal does not include multiple server failover. Only the single failover of the
SB1, SB2 and SB3 SRMGCs is provided.
1.5 Session Initiated Protocol (SIP)
VoiceCon requires that the proposed IPTS support SIP-compatible stations
and trunk networking as specified in the most current IETF Work Group RFC
document. It is also required that the IPTS solution be capable of supporting
the IETF-sponsored signaling protocols used for Internet conferencing,
telephony services and features, presence, events notification and instant
messaging.
1.5.1 SIP Stations
Vendor Response Requirements
Indicate if the IPTS solution as proposed can currently support SIPcompatible desktop telephone instruments (self and/or third party) and PC
client softphones assuming 20% of individual system end user stations are
IP-based. Specify if SIP call control is embedded in the IPTS common control
design or optional hardware/software elements are required. Also identify up
to three (3) third party SIP telephones you have successfully tested for
operation behind your proposed IPTS solution.
NEC Response: The Univerge SV7000-S Module supports SIP communications to NEC
and other SIP-compatible telephones and PC Client Softphones. The SV7000-S has
other functions also and so is always included in the SV7000 system. NEC has tested
the NEC ITN SIP telephone and the Polycom SIP Telephones.
1.5.1.1 SIP Clients
Vendor Response Requirements
Do the IP desktop telephone instruments and PC client softphones included
as part of this proposal in response to RFP Section 5: Voice Terminals
currently conform to IETF SIP standards? If not, are they upgradeable to
support SIP standards and specifications via a firmware download if required
in the future? If a firmware download is required is there an associated cost
or fee to VoiceCon?
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NEC Response: The NEC Dterm IP telephones and the SP-30 Softphone do not
conform to the SIP standards. The Dterm IP telephones can be converted to ITN
telephones with SIP capabilities by firmware download. It would also be necessary to
convert the Dterm IP license to a SIP license on the SV7000. If the system was to be
ordered with ITN telephones, the total price may actually be less. But if the firmware
conversion is done after cutover, there will be a processing fee.
1.5.2 SIP Trunk Networking
Vendor Response Requirements
Indicate if the IPTS network solution as proposed can support SIP-based
trunk networking. Specify if SIP media proxies are required to support this
requirement. Identify up to three (3) major Service Providers (SPs) and three
(3) other IPTS suppliers you have conducted IP trunk networking
compatibility tests with for the proposed IPTS.
NEC Response: The NEC Univerge SV7000 does not currently support SIP-based
trunk networking. NEC is willing to work with other vendors to explore this possibility.
1.5.2.1 SIP Applications
Vendor Response Requirements:
Indicate if the IPTS network solution as proposed can support SIP-enabled
applications, such as Internet conferencing, telephony services and features,
presence, events notification and instant messaging.
NEC Response: No, the NEC Univerge SV7000 does not currently support 3rd party SIP
applications. However, NEC does make applications of this type. For example, NEC’s
Personal Call Assistant (PCA) can be configured onto every PC and integrate the PC
with the telephone to allow one-click dialing, voice announce of incoming calls,
presence, and automatic forwarding based on calendar entries in Outlook.
1.6 Security
VoiceCon requires a secure IPTS network solution to optimize system
performance and reduce the probability of toll fraud and illegal system
access.
1.6.1 Authentication
Vendor Response Requirements
Briefly describe authentication processes embedded in the proposed IPTS
solution to prevent: unauthorized access to common control elements, data
resources; and abuse of telephony services, e.g., toll fraud.
NEC Response: The Univerge SV7000 does not have an open user direct interface as
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might be found on a server-based system. The SV7000 only communicates with
telephones and gateways that are registered by MAC address. The programming
interface is a connection through the MA4000 which has a built-in set of features (called
CAS, Customer Authentication System) which protect against unauthorized access. Toll
Fraud is a potential problem with any telephone system, IP or TDM. NEC has
successfully employed a myriad of features for Toll Fraud protection in both TDM and IP
system for many years. All of the Toll Fraud prevention features of the NEC TDM
telephone systems are available in the SV7000 including: Least Cost Routing, 3 and 6digit screening, Authorization Codes, Account Codes, Number Blocking and Outgoing
Trunk Restrictions.
1.6.2
Disruption of Services
Vendor Response Requirements
Briefly describe any embedded features/functions in the proposed IPTS that
will reduce probability of telephony services disruption due to Denial-ofService (DoS) attacks.
NEC Response: None, this is a LAN/WAN issue and needs to be addressed in the
LAN/WAN configuration and programming.
1.6.3
Confidentiality and Privacy (Packet Sniffing)
Vendor Response Requirements
Briefly described any embedded features/functions in the proposed IPTS that
will preserve communications confidentiality and privacy. Indicate if control
signaling and/or bearer communications signaling is encrypted at the call
control, voice client, and media gateway elements to counter packet sniffing
attempts.
NEC Response: Yes, both the signaling and the voice packets are encrypted to prevent
packet sniffing.
1.6.4
Physical Interfaces
Vendor Response Requirements
Are there separate physical network interfaces to IPTS administration,
control, and voice transmission signaling functions?
NEC Response: No, all connections to the SV7000 are made through the single RJ-45
connector to the LAN. There is a terminal control port on the back panel of the SV7000
for console operation. It is password protected.
1.6.5
Root Access
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Vendor Response Requirements
Is there direct Root access to the IPTS common control?
NEC Response: No, the SV7000 runs an NEC-proprietary Operating System to prevent
such access.
2.0 IPTS Network Port Capacity Requirements
The proposed IPTS must be capable of supporting port capacity requirements
for the HQ facility and remote branches. It must also be capable of
supporting future VoiceCon growth requirements at HQ and RO facilities.
2.1.0 Port Capacity Requirements
The equipped port capacity of the proposed VoiceCon HQ IP Telephony
System at time of installation and cutover must support of a mix of IP
telephones, analog telephones, facsimile terminals, modems, local central
office trunk circuits (analog and digital, long distance trunk circuits [digital,
only], and private network trunk circuits [IP]).
In support of general communications requirements, VoiceCon facilities will
have a sufficient number of wiring closets distributed throughout each facility
to satisfy ANSI/EAI/TIA 569 structured cabling specifications for voice and
data communications. Wiring closets will be interconnected based on
requirements of the selected system. The entrance facility (trunk connect
panel), main telecom equipment room, and Main Distribution Frame (MDF)
for each facility are located off the entrance lobby. It will be the
responsibility of the contractor to provide all cross connects between labeled
110 terminal blocks in each wiring closet and the demarc or "smart jack" and
their equipment. The following sections describe the port capacity
requirements for each of the VoiceCon network locations. Satisfying these
stated port capacity requirements is a MANDATORY requirement
2.1.1
HQ Facility
The HQ location is a four-floor facility that will support at time of system
installation and cutover the following station equipment:
* 1040 desktop IP station instruments;
* 100 PC client softphones (including three attendant operator positions);
* 10 IP audio conferencing units;
* 26 analog telephones including 5 used for Power Failure Transfer Station
operation;
* 12 facsimile terminals;
* 12 data modems.
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Station equipment is uniformly distributed within and across the four floors of
the building. There are ten (10) wiring closets per floor, and one (1) main
equipment room on the first floor. See Table 1 for station summary.
2.1.2 RO Facility
The RO facility will be a two floor facility that will support:
* 205 desktop IP station instruments;
* 21 PC client softphones (including two (2) attendant operator positions);
* 4 IP audio conferencing units;
*10 analog telephones including 2 used for Power Failure Transfer Station
operation;
* 5 facsimile terminals;
* 5 data modems.
Station equipment is uniformly distributed within and across the two floors of
the building. There are ten (10) wiring closets per floor, and one (1) main
equipment room on the first floor. See Table 1 for station summary.
2.1.3 SB1 (Large)
The SB1 facility will be a single floor facility that will support:
* 75 desktop IP station instruments
* 10 PC client softphones
* 4 audioconferencing units;
* 5 analog telephones including 2 Power Failure Transfer Stations;
* 2 facsimile terminals;
* 4 modems.
All line station equipment will be equally distributed across the single floor of
the building. There will be two (2) wiring closets, and one (1) main
equipment room. See Table 1 for station summary.
2.1.4 SB2 (Medium)
The SB1 facility will be a single floor facility that will support:
* 37 desktop IP station instruments;
* 5 PC client softphones
* 2 IP audioconferencing units;
* 2 Analog station used as Power Failure Transfer Stations;
* 2 facsimile terminals;
* 2 modems.
All line station equipment will be equally distributed across the single floor of
the building. There will be two (2) wiring closets, and one (1) main
equipment room. See Table 1 for station summary.
2.1.5 SB3 (Small)
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The SB3 facility will be a single floor facility that will support:
* 8 Deskop IP station instruments;
* 1 Analog station used as a Power Failure Transfer Station;
* 1 facsimile terminal.
All station equipment will be equally distributed across a single room on the
main floor of the building. There will be one (1) wiring closet/equipment
room. See Table 1 for station summary.
Table 1: VoiceCon Equipped Station Requirements
IP Station
IP Station
IP Att.
Analog
Analog
Fax
Modem
Softconsole
IP
AudioConf.
Instrument
Softphone
Standard
PFTS
Terminal
Device
HQ
1040
97
3
10
21
5
12
12
RO
205
19
2
4
8
2
5
5
SB1
75
9
1
4
3
2
2
4
SB2
37
5
0
2
0
2
2
2
SB3
8
0
0
0
0
1
1
0
2.2 Equipped Voice Terminal Requirements
VoiceCon requires the following mix of wired and installed desktop IP
telephone instruments (Table 4). Note: Descriptions of Desktop IP individual
voice terminal types can be found in RFP Section 4
Table 4: VoiceCon Desktop IP Telephone Instruments
Requirements
Facility
Economy
Administrative
Professional
Executive
HQ
50
140
800
50
RO
15
30
150
10
SB1
8
10
55
2
SB1
3
8
25
1
SB3
0
1
7
0
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Trunk Circuit Requirements
The VoiceCon HQ and RO facilities will each have a combination of local, long
distance, and private network trunk circuits. The SB facilities will each have a
limited number of analog trunk circuits, but all long distance calls will be
routed through the HQ facility. All facilities will also have PFTS circuits. All
local digital trunks must be able to support a combination of inbound DID
service and two-way CO trunk services. All long distance calls placed from a
SB facility will be routed via the LAN/WAN through the HQ facility for PSTN
trunk access.
The following table summarizes HQ facility trunk circuit requirements for
each of the four VoiceCon design configurations.
Table 5: VoiceCon IPTS Network Equipped Trunk Port Requirements
Per
Incremental
Location
T-1 Digital Local
Inbound/Outbound
T-1 Digital
Long
Distance
Analog
(PFTS)
2-way GS/LS
HQ
6
7
5
25
RO
2
2
2
10
SB1
1
0
2
5
SB2
0
0
2
10
SB3
0
0
1
5
VoiceCon will engineer its WAN trunk circuits to support compressed voice
traffic (G.729A algorithm voice codecs) among all IPTS network facilities
inter-facility voice traffic. Any additional PSTN trunk circuits required to
support local survivability requirements must be identified.
Necessary common equipment must be included in the system
configuration and pricing proposals and identified as such.
Vendor Response Requirements
Confirm that the proposed IPTS network solution satisfies the stated trunk
circuit requirements; support of centralized long distance trunk resources at
the VoiceCon HQ facility for the SB facilities; and automatic alternate routing
of calls among all VoiceCon facilities across the WAN and PSTN.
NEC Response: The proposed NEC Univerge SV7000 system satisfies the stated
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station and trunk circuit requirements of the above three tables. The station and trunk
distribution will be as specified above and provide for a centralized long distance trunk
resource at the HQ with local trunk facilities at all five locations. Any station at any
location can, with sufficient class of service, make an outgoing call through any other
location’s local trunks so as to save on Telco charges. This is done automatically using
the Least Cost Routing features of the SV7000. Incoming calls can be received at any
location with a destination of any station on the network. These calls may be routed
directly via DID functionality or they may be routed to an attendant for service.
Additionally, the call may be automatically forwarded from the destination station to
voicemail if the station is busy or does not answer.
2.3.1 ISDN PRI Services
All installed VoiceCon T-1 trunk circuits must support ISDN PRI features and
functions for both local and long distance exchange carrier transmission
services.
Vendor Response Requirements
Confirm that the proposed T-1 trunk circuit interfaces support ISDN PRI
capabilities.
NEC Response: The proposed gateways for T1 circuits will all support ISDN PRI
circuits and all of the capabilities supported on those circuits, such as DID, two-way
calling, ANI, E911, etc.
2.4
IPTS Network Growth Requirements
VoiceCon anticipates that station capacity requirements at the HQ and RO
facilities will increase approximately 50% for the expected installed life of the
proposed IPTS network solution. Port capacity growth requirements at the
SB1 and SB2 faciilties are anticipated to increase by about 20%; no growth is
anticipated at the SB3 facility.
Vendor Response Requirement:
Confirm that the proposed IPTS solution can satisfy VoiceCon station port
growth requirements and associated trunk growth requirements at its HQ,
RO, SB1 and SB2 facilities without replacing any hardware equipment at time
of initial system installation and cutover. Hardware additions are permissible
to support incremental port interface requirements.
NEC Response: The SV7000 network can grow from a minimum configuration of less
than 100 ports to a maximum configuration of 192,000 ports without discarding or
replacing any hardware. This modular growth is a feature NEC has been providing for
over 20 years on telephony systems and we have carried it forward into the SV7000.
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3.0 Port Interface and Traffic Handling Requirements
The proposed IPTS network solution must be able to support a mix of
TDM/PCM and IP ports. For traffic design engineering calculations assume
the following traffic requirements:
1. The average busy hour traffic for IP desktop station users will be rated
at 10 CCS @ P.01. Assume a traffic mix pattern of 30% intra-network
calls, 15% outgoing local trunk calls, 25% outgoing long distance
trunk calls, and 30% incoming DID trunk calls.
2. Analog telephone station busy hour traffic will be rated at 3 CCS @
P.01. Assume a traffic mix pattern of 70% inter-network calls and
30% outgoing local trunk calls. All analog telephone station calls will
be subject to toll restrictions.
3. Assume that busy hour traffic is rated at 36 CCS @P.01 for each of the
following port types: all PSTN and WAN trunk circuits; attendant
consoles; modems; audioconferencing units; facsimile terminals; voice
mail ports.
Vendor Response Requirement:
The proposed system must design and engineer their system to support the
above traffic assumptions. Confirm you have satisfied this requirement.
NEC Response: The proposed system is fully IP connected and therefore is 100% nonblocking except when the LAN/WAN becomes congested and causes blocking. If a
particular end-device (station of trunk) is idle, then any other idle station or trunk can
connect to it. This means that there are a full 36 CCS available on every port on the
system.
3.1 Circuit Switched Network Design
The proposed IPTS solution must support a variety of peripheral ports and
switched connections. Although it is not required to support traditional
digital voice terminal equipment, the IPTS must support analog
communications devices. Switched connections involving non-IP ports may
be handled using a circuit switched network, media gateways/Ethernet
switches, or a combination of both methods.
Vendor Response Requirement:
If the proposed IPTS network solution includes integrated circuit-switched
hardware equipment, then briefly describe the characteristics of the offering.
Include, at minimum, the following information: hardware cabinet
description; CCS @P.01 rating; center stage switch and local TDM bus time
slot/talk slot capacities; interswitch link capacities; all redundant design
elements and level of redundancy.
NEC Response: The Univerge SV7000 proposed for this RFP will be configured with no
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TDM devices except the gateways between IP and TDM circuits. All switching is done on
the IP network and therefore no Time Division Switch has been included.
3.2 Peer-to-Peer Communications for IP Station to IP Station Calls
All two-party voice calls between IP desktop stations located at VoiceCon
facilities must be handled exclusively over the LAN/WAN infrastructure
without any circuit switched connections. This is a Mandatory requirement.
Vendor Response Requirement:
Confirm that your proposed system satisfies this Mandatory requirement.
NEC Response: Confirmed. NEC’s Peer-to-peer switching means the stations
participating in a call are connected directly to each other through the IP network. The
voice signals travel through the IP network but do not "go through" the switch as they do
in traditional telephony.
3.2.1
IP Station Discovery
How do IP communications devices learn about their voice VLAN, including IP
addresses, default gateways, call controller, TFTP server, QoS settings,
VLANs, and other parameters. Does the proposed system solution employ
proprietary protocols for IP communications devices to learn their voice VLAN
or is an industry standard, such as Dynamic Host Control Protocol (DHCP)
used?
3.2.2
IP Station Power over Ethernet (PoE)
VoiceCon requires that the power option to support IP telephones conform to
IEEE 802.3af Power over Ethernet (PoE) standards.
Vendor Response Requirement:
Confirm that the proposed IPTS solution supports the IEEE 802.3af
specification for in-line of IP telephone equipment. Describe current, future
and retrospective compatibility of all proposed equipment. If 802.3af is not
supported, identify the PoE implementation being proposed.
NEC Response: The NEC Univerge SV7000 Dterm IP telephones support IEEE 802.3af
Power over Ethernet specifications.
3.2.3 IP Station QoS
Vendor Response Requirement:
Describe the proposed IPTS solution’s capabilities to provide Layer 2 and
Layer 3 QoS to IP stations to ensuring end-to-end quality of service. Include
in the response what industry standards are deployed.
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NEC Response: The NEC Univerge SV7000 provides Layer 2 and 3 QoS to IP stations
through VLAN tagging IEEE 802.3 p and q. By implementing VLANs, the IP terminal can
easily determine priority requirements between the packets from the voice system and
those for the attached data terminal.
3.3 Multi-Party Conference Calls
The proposed system must be able to support six party add-on conference
calls among IPTS stations and off-network stations. The system must also
support a minimum of three (3) off-network stations per multi-party
conference call when required. The HQ IPTS must support a minimum of 20
simultaneous multi-party add-on conference calls (up to six parties per
conference) and the RO IPTS a minimum of 10 simultaneous multi-party addon conference calls (up to six parties per conference)
Vendor Response Requirement:
Briefly explain how multi-party add-on conference calls are handled if:
1) All parties are on-network IP stations;
2) There is a mix of on-network IP and off-network stations.
The explanation should identify any and all hardware and software
requirements necessary to support multi-party add-on conference call
requirements. Specify if peripheral hardware equipment, e.g., conference
bridge servers, is required.
NEC Response: The Univerge SV7000 offers pure IP conferencing using the VS-32
Conference Server. Any conference call must contain at least one internal party, but all
others may be either internal or external. The VS32 conferencing server supports 3, 8,
16 and 32 party conferences for peer-to-peer ports. To meet the RFP requirements,
each node was configured with a number of 8-party conferences equal to the number of
6-party conferences required. Then the correct number of VS-32 servers was configured
with each server providing four 8-party conferences.
3.4 VoIP Overflow Traffic
If available WAN circuits connecting the HQ, RO and all SB facilities are busy,
call admission control levels are reached, or QoS levels are not satisfied onnetwork voice traffic must be able to automatically overflow to PSTN trunk
circuits.
Vendor Response Requirement:
Confirm that your proposed communications system supports overflow of
voice traffic across VoiceCon locations if WAN links are not available or
conditions are not acceptable. Also indicate if overflow traffic can revert back
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to the WAN if conditions permit.
NEC Response: Yes, the NEC Univerge SV7000 supports overflow of network traffic
between nodes to the PSTN if the WAN is busy by using the Least Cost Routing
features of the SV7000. When the WAN is freed up, new calls will route via the WAN
first. However, calls in process will not reroute between the WAN and the PSTN
automatically.
3.5.0 Port Interface Circuit Cards
For each of the following port types, provide a brief description of the
proposed port interface circuit card(s) and/or media gateway equipment
included with the proposed IPTS to support analog, digital, and IP ports.
Include in the descriptions below the number of port interface terminations
for each port circuit card, and the number of available gateway channels for
each media gateway unit.
3.5.1 IP Telephones (desktop instrument and PC client softphones,
including Attendant Console Position) & IP Audioconferencing Units
Vendor Response Requirement:
Provide a brief description how all IP telephone types are logically and
physically supported by the common control call telephony server. If direct
call control signaling via the Ethernet LAN/WAN is not supported identify all
intermediary carrier, signaling interface and/or media gateway equipment
that is required.
NEC Response: The SV7000 common control software supports the IP Dterm phones
over the IP network. The IP Dterms connect directly to the network and communicate
with the SV7000 for control signaling only. IP stations communicate with each other
through peer-to-peer protocols for all voice transmissions.
3.5.2 Analog telephones
Vendor Response Requirement:
Provide a brief description how analog telephones are logically and physically
supported by the common control call telephony server, identifying all
intermediary hardware elements necessary for control signaling transmission.
Specify the number of circuit terminations per circuit board/module/media
gateway.
NEC Response: At all locations, the MPS Gateway with the SCA-8LC Module provides
the interface function between the IP network (LAN) and conventional terminals, such as
analog telephones and G3 FAX machines. The Gateway can be used as a gateway
device for mutual converting between voice IP packet and TDM-based voice signals.
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3.5.3 Facsimile terminal
Vendor Response Requirement:
Provide a brief description how facsimile terminals are logically and physically
supported by the common control call telephony server, identifying all
intermediary hardware elements necessary for control signaling transmission.
Specify the number of circuit terminations per circuit board/module/media
gateway.
NEC Response: At all locations, the MPS Gateway with the SCA-8LC Module provides
the interface function between the IP network (LAN) and conventional terminals, such as
analog telephones and G3 FAX machines. The Gateway can be used as a gateway
device for mutual converting between voice IP packet and TDM-based voice signals.
3.5.4. Modem
Vendor Response Requirement:
Provide a brief description how modem terminals are logically and physically
supported by the common control call telephony server, identifying all
intermediary hardware elements necessary for control signaling transmission.
Specify the number of circuit terminations per circuit board/module/media
gateway.
NEC Response: At all locations, the MPS Gateway with the SCA-8LC Module provides
the interface function between the IP network (LAN) and conventional terminals, such as
analog telephones and modems. The Gateway can be used as a gateway device for
mutual converting between voice IP packet and TDM-based voice signals.
3.5.5 Power Failure Transfer Station (PFTS)
Vendor Response Requirement:
Provide a brief description how analog telephone instrument Power Failure
Transfer Stations (PFTSs) are logically and physically supported by the
common control call telephony server, identifying all intermediary hardware
elements necessary for control signaling transmission. Specify the number of
circuit terminations per circuit board/module/media gateway.
NEC Response: The Survivable Remote-Media Gateway Controller (SR-MGC) is
placed at the remote office to manage all calls to/from IP terminals and trunk calls
to/from PSTN via the Media Gateway in case of network failure or MGC breakdown at
the HQ office. In normal state, SR-MGC operates in stand-by mode under monitoring of
the main office MGC. Once the HQ MGC cannot accept registration of IP terminals in the
remote office, the operation is switched over to the SR-MGC. When the main office MGC
recovers, the SR-MGC returns to stand-by mode again.
NEC has also provided Power Failure Transfer (PFT) at the remote locations through the
use of the MPS Gateway SCA-6COT Module with integrated PFT. If power or control via
IP is lost, the Gateway goes into PFT mode. In this mode the one CO trunk connection
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and one of the analog station ports are directly connected. When power or control is
restored, calls in progress continue until completion and the ports then reset to normal
operation.
3.5.6 GS/LS CO Trunk
Vendor Response Requirement:
Provide a brief description how GS/LS CO trunk circuits are logically and
physically supported by the common control call telephony server, identifying
all intermediary hardware elements necessary for control signaling
transmission. Specify the number of circuit terminations per circuit
board/module/media gateway.
NEC Response: The MPS Gateway SCA-6COT Module provides 6 CO Trunk
Connections.
3.2.7 DS1/T-1 Carrier Interface Trunk
Vendor Response Requirement:
Provide brief description how DS1-based T-1 carrier trunk circuits are
logically and physically supported by the common control call telephony
server, identifying all intermediary hardware elements necessary for control
signaling transmission. Specify the number of circuit terminations per circuit
board/module/media gateway.
NEC Response: The Digital MG-PRI and the MPS Gateway SCA-PRI provide the ISDN
Gateway function between the IP network (LAN) and the ISDN network (PRI). Each
gateway supports a single T1 circuit with ISDN PRI protocol.
3.2.8 Other Trunk Interfaces
VoiceCon may need at some future time additional analog trunk interfaces,
specifically Auxiliary, FX, and E&M Tie Line.
Vendor Response Requirement:
Provide a brief description of how additional analog trunk interface
requirements can be logically and physically supported by the common
control call telephony server, identifying all intermediary hardware elements
necessary for control signaling transmission. Specify the number of circuit
terminations per circuit board/module/media gateway.
NEC Response: The MPS Gateway SCA-6COT Module provides 6 CO Trunk
Connections. FX lines are supported from the SCA-8LC Module using one of the 8
analog station ports. E&M Tie Lines are not supported directly but can be routed through
the ISDN PRI.
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4.0.0. Voice Terminal Instruments
The proposed communications system must be able to support a mix of
analog and IP communications devices. VoiceCon will provide its own analog
telephone instruments, fax terminals, and modems.
4.1 Regulation Requirements
All single- and multi-line IP phones will be manufactured in accordance with
Federal Communication Commission hearing aid compatibility technical
standards contained in Section 68.316. and the Telecommunication Act of
1996.
Vendor Response Requirement:
Confirm the proposed telephone equipment satisfies these requirements
NEC Response: Confirmed. All NEC current release telephones are in compliance with
the FCC regulations for hearing aid compatibility.
4.2 Desktop IP Telephone Instruments
VoiceCon has a requirement for several types of desktop IP telephone
instruments:
•
•
•
•
4.2.1
Economy
Administrative
Professional
Executive
Economy Desktop IP Telephone Instrument
The Entry model will be used in common areas. It should have, at minimum,
the following design attributes and features/functions:
•
•
•
•
•
•
•
12 key dial pad
Single line appearance
Hold button
G.711/G.729 voice codecs
Auto Self Discovery/DHCP
Echo Canceller
IEEE 802.af POE support
Vendor Response Requirement:
Confirm that your proposed Economy model satisfies all of the stated
requirements and provide a brief product description that includes an
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illustration/ photograph (PPT format, only) of the instrument. Indicate in your
response any and all requirements not satisfied.
NEC Response: NEC has chosen the Dterm IP ITR-4D telephone to meet this
requirement. The ITR-4D is a 4-button display telephone that meets all of the
requirements stated above. Additionally, there will be three feature programming keys
which may be used to program either feature access of speed dial (frequently dialed)
numbers.
The proposed Dterm IP Version 3 terminals include the following features:
Four Local Soft Key Controls (detail functions are dependent on PBX)
Message Waiting LED
24 Character, 3-Line LCD
Speed Dial/DSS Buttons
Programmable Line Keys to support almost every feature/function that the
TDM Dterm Series i terminal supports
11 Dedicated Function Keys (Feature, Recall, Conf, Redial, Hold, Transfer,
Answer & Speaker). All Dterm IP models with the exception of the 4D Dterm
IP phone which has 9 Dedicated Function Keys.
12 Key Dial Pad
Two 10/100 full duplex Ethernet ports - One connects the Dterm IP to the
local Ethernet network; the other provides connectivity for a local workstation.
Powering options
•
Local AC adapter (optional hardware)
•
In-line power support utilizing the 802.3af protocol
Transportable QoS follows the user regardless of log-in location.
Multiple Voice Coding support automatically negotiates to a common setting.
Additionally, NEC is including the Personal Call Assistant (PCA) OAI application for all IP
terminals to provide desktop access to a 99 call inbound calls log, a 99 call outbound
calls log, a corporate directory, a personal directory, and a telephone user status (in
meeting, on travel, etc.) application. This PCA is visible on the PC associated with the
telephone.
The Dterm IP 4-line Display phone is being proposed for the single line IP telephone
requirement.
Terminal Attribute (Dterm 4-Line
Display)
VOICE CODECS:
G.711
G.729A
Requirement
Supported
Yes
Yes
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VoiceCon Spring 2007
AUTO SELF DISCOVERY/DHCP
INTEGRATED ETHERNET SWITCH
(2 ports)
INTEGRATED THIN CLIENT
BROWSER FUNCTION (HTML,
XML, WAP, other)
ECHO CANCELER
QoS SUPPORT
802.1p/Q
DIFFSERV
# PROGRAMMABLE
LINE/FEATURE KEYS (MINIMUM)
VISUAL LINE/FEATURE KEY
ACTIVITY INDICATOR
FIXED FEATURE BUTTONS
HOLD
RELEASE
LAST NUMBER REDIAL
TRANSFER
SPEAKER/MUTE
VOLUME CONTROL
MESSAGE WAITING INDICATOR
TWO-WAY FULL DUPLEX
SPEAKERPHONE
DISPLAY FIELD (2 lines x 24
characters, min)
LARGE PIXEL DISPLAY FIELD (8
lines x 40 characters, minimum)
STORED CALL DATA
LAST INCOMING CALLS (10,
minimum)
LAST NUMBERS DIALED (10,
minimum
INTEGRATED PBX SYSTEM
DIRECTORY ACCESS W/DIAL BY
NAME
LDAP SERVER DIRECTORY
ACCESS
POWER over ETHERNET
Request for Proposal
for an IP Telephony System
Yes
Yes
N/A
Yes
Yes
Yes
Yes
4
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
N/A
PCA
PCA
PCA
PCA
PCA
Yes
A picture of this terminal is included in the attached Dterm PowerPoint file.
4.2.2
Administrative Desktop IP Telephone Instrument
The Administrative model will be used by station users who have executive
management group call answering and coverage responsibilities. It should
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have, at minimum the following design attributes and features/functions:
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
12 key dial pad
Sixteen (16) programmable line/feature keys with soft label/status
indicators
G711, G729 and wideband, e.g., G.722, voice codecs
Auto Self Discovery/DHCP
Echo Canceller
QoS Support (802.1p/Q, DiffServ)
Hold key
Last Number Redial key
Release key
Message Waiting/Call Ringing indicator(s)
Full Duplex Speakerphone
Speaker/Mute key
Volume Control keys/slide
High resolution, backlit, monochrome grayscale pixel-based, graphical
display screen with four (4) associated context sensitive soft feature
labels (key, cursor, or navigator control)
LDAP access
Stored Call Data (Last 10 numbers dialed/Last 10 incoming call
numbers)
Integrated Ethernet switch and two (2) RJ-45 connector interface ports
for 10/100 Mbps connectivity
Headset interface
IEEE 802.af POE support
The Administrative model must also be capable of supporting optional add-on
key modules if an additional 12 programmable line/feature keys with soft
label/indicator status is required at some future time.
Vendor Response Requirement:
Confirm that your proposed Administrative model satisfies all of the stated
requirements. Provide a brief product description that includes an
illustration/photograph (PPT format, only) of the instrument. Indicate in your
response any and all requirements not satisfied. State which required
feature-specific keys are not available, but softkey feature access can be
used as an alternative.
NEC Response: NEC has chosen the Dterm IP 240G INASET telephone set to meet
this requirement.
The INASET 240G offers outstanding functionality, intuitive user interfaces, crystal clear
audio quality, stylish design, network manageability and state-of-the-art features. The
entry-level INASET was designed for companies with converged IP infrastructures that
need to deliver productivity-enhancing applications and services directly to the desktop.
An enterprise-class IP terminal, the INASET 240G has 3” LCD display and 16
programmable keys and softkeys for access to applications such as unified messaging,
company directories and call logs. The applications can be custom-designed or are
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provided by the NEC Openwork® package.
Key Features
The INASET line supports a comprehensive suite of advanced applications for mobility
and collaboration and is especially well-suited for retail, hospitality, education and
government. The INASET 240G is one in a series of NEC IP terminals designed to meet
the diverse communication needs of users across the enterprise - from affordable entrylevel IP phones to sophisticated network appliances. All are designed with ergonomics
and modern office aesthetics in mind.
The NEC INASET 240 G is designed with conveniently placed, easy-touse features.
Pixel-based display—A gray scale 240 x 160 pixel display provides intuitive
access to calling features.
Four softkeys access various services through an external XML server. Here are
some of the basic services provided with the integration of an OpenWorX server
Software Development Kit allows the development of third-party XML-based
applications using the higher level ASP/.NET wrapper or the native INASET XML
specification.
Built-in speaker allows for hands-free operation.
Full duplex audio support when the INASET is set for hands-free mode.
Microphone port provides support for the external NEC microphone to be used in
a hands-free mode.
Built-in 3-port switch provides automatic negotiation for 10/100Mb network
connection.
16 programmable multi-function keys can be set to access any of the hundreds
of terminal features within the NEAX PBX.
Six function keys fixed to the base of the terminal: Hold, Directory, Message,
Speaker, Microphone and Transfer.
Support for 7Khz of audio when connected to another1 INASET product.
Support for a multitude of powering options: Local power, Ethernet power via
802.3af or Cisco Discover Protocol. Also supports the NEC discovery protocol
with the ILPA dongle integration.
The INASET 240G also provides an expansion port that allows an
additional feature to be driven at the desktop:
With the integration of the AD (A) adapter, you can now equip your desktop
terminal for external recording capabilities. Incoming and outgoing calls can be
recorded to a local PC hard drive or tape recorder which is connected to the
INASET 240G.
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VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
Or add the PS (A) adapter to the INASET 240G which provides survivability to
the desktop. If the IP network crashes, your INASET has a backup PSTN circuit
that provides incoming and outgoing call capabilities. This is a perfect add-on for
users who require INASET power but are located miles or even hundreds of
miles away from their servicing switch.
Terminal Attribute (Dterm IP 240G INASET)
Requirement
Supported
12 key dial pad
Sixteen (16) programmable line/feature keys
with soft label/status indicators
Voice Codecs:
G.711
G.729A
Wideband
Auto Self Discovery/DHCP
Echo Canceler
QoS Support
802.1p/Q
DiffServ
Hold Key
Last Number Redial Key
Release Key
Message Waiting / Call Ringing Indicator
Two-Way Full Duplex Speakerphone
Speaker/Mute Key
Volume Control Keys/Slide
High resolution, backlit, monochrome
grayscale pixel-based, graphical display
screen with four (4) associated context
sensitive soft feature labels (key, cursor, or
navigator control)
Yes
Yes, shown 8 at a time
on the display
LDAP Acess
Stored Call Data
Last Incoming Calls (10, Minimum)
Last Numbers Dialed (10, minimum
Integrated Ethernet switch and two (2) RJ-45
connector interface ports for 10/100 Mbps
connectivity
Integrated PBX System Directory Access
W/Dial by Name
Headset Interface
Power over Ethernet
PCA
PCA
PCA
PCA
Yes
Yes
Yes
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
PCA
Yes
Yes
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VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
A picture of this terminal is included in the attached Dterm PowerPoint file.
4.2.3 Professional Desktop IP Telephone Instrument
The Professional model will be used by VoiceCon managers. It should have,
at minimum the following design attributes and features/functions:
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
12 key dial pad
Six (6) programmable line/feature keys with soft label/status
indicators
G711, G729 and wideband voice codecs
Auto Self Discovery/DHCP
Echo Canceller
QoS Support (802.1p/Q, DiffServ)
Hold key
Last Number Redial key
Release key
Message Waiting/Call Ringing indicator(s)
Full Duplex Speakerphone
Speaker/Mute key
Volume Control keys/slide
High resolution, backlit, monochrome grayscale pixel-based, graphical
display screen with four (4) associated context sensitive soft feature
labels ((key, cursor, or navigator control)
LDAP access
Stored Call Data (Last 10 numbers dialed/Last 10 incoming call
numbers)
Integrated Ethernet switch and two (2) RJ-45 connector interface ports
for 10/100 Mbps connectivity
Bluetooth interface for wireless headset
USB interface
IEEE 802.af POE support
The Professional model must also be capable of supporting the following
integrated feature/functions if required at some future time:
•
•
Gigabit (10/100/1000 Mbps) Ethernet connectivity
Embedded Web-browser applications
Vendor Response Requirement:
Confirm that your proposed Professional model satisfies the stated
requirements and provide a brief product description that includes an
illustration or photograph (PPT format, only) of the instrument. Indicate in
your response any and all requirements not satisfied. State which required
feature-specific keys are not available, but softkey feature access can be
used as an alternative.
NEC Response: NEC has chosen the Dterm IP 320G INASET telephone set to meet
Page 44
VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
this requirement.
The INASET 320G offers outstanding functionality, intuitive user interfaces, crystal clear
audio quality, stylish design, network manageability and state-of-the-art features. The
INASET was designed for companies with converged IP infrastructures that need to
deliver productivity-enhancing applications and services directly to the desktop. An
enterprise-class IP terminal, the INASET 320G has 4” LCD display and 16
programmable keys and softkeys for access to applications such as unified messaging,
company directories and call logs. The applications can be custom-designed or are
provided by the NEC Openwork® package.
Key Features
The INASET line supports a comprehensive suite of advanced applications for mobility
and collaboration and is especially well-suited for retail, hospitality, education and
government. The INASET 320G is one in a series of NEC IP terminals designed to meet
the diverse communication needs of users across the enterprise - from affordable entrylevel IP phones to sophisticated network appliances. All are designed with ergonomics
and modern office aesthetics in mind.
The NEC INASET 320 G is designed with conveniently placed, easy-touse features.
Pixel-based display—A gray scale 320 x 160 pixel display provides intuitive
access to calling features.
Four softkeys access various services through an external XML server. Here are
some of the basic services provided with the integration of an OpenWorX server
Software Development Kit allows the development of third-party XML-based
applications using the higher level ASP/.NET wrapper or the native INASET XML
specification.
Built-in speaker allows for hands-free operation.
Full duplex audio support when the INASET is set for hands-free mode.
Microphone port provides support for the external NEC microphone to be used in
a hands-free mode.
Built-in 3-port switch provides automatic negotiation for 10/100Mb network
connection.
16 programmable multi-function keys can be set to access any of the hundreds
of terminal features within the NEAX PBX.
Six function keys fixed to the base of the terminal: Hold, Directory, Message,
Speaker, Microphone and Transfer.
Support for 7Khz of audio when connected to another1 INASET product.
Support for a multitude of powering options: Local power, Ethernet power via
802.3af or Cisco Discover Protocol. Also supports the NEC discovery protocol
with the ILPA dongle integration.
Page 45
VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
The INASET 320G also provides an expansion port that allows an
additional feature to be driven at the desktop:
With the integration of the AD (A) adapter, you can now equip your desktop
terminal for external recording capabilities. Incoming and outgoing calls can be
recorded to a local PC hard drive or tape recorder which is connected to the
INASET 320G.
Or add the PS (A) adapter to the INASET 320G which provides survivability to
the desktop. If the IP network crashes, your INASET has a backup PSTN circuit
that provides incoming and outgoing call capabilities. This is a perfect add-on for
users who require INASET power but are located miles or even hundreds of
miles away from their servicing switch.
Terminal Attribute (Dterm IP 320G INASET)
12 key dial pad
Sixteen (16) programmable line/feature keys
with soft label/status indicators
Voice Codecs:
G.711
G.729A
Wideband
Auto Self Discovery/DHCP
Echo Canceler
QoS Support
802.1p/Q
DiffServ
Hold Key
Last Number Redial Key
Release Key
Message Waiting / Call Ringing Indicator
Two-Way Full Duplex Speakerphone
Speaker/Mute Key
Volume Control Keys/Slide
High resolution, backlit, monochrome
grayscale pixel-based, graphical display
screen with four (4) associated context
sensitive soft feature labels (key, cursor, or
navigator control)
LDAP Acess
Stored Call Data
Last Incoming Calls (10, Minimum)
Last Numbers Dialed (10, minimum
Requirement
Supported
Yes
Yes, shown 8 at a time
on the display
Yes
Yes
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
PCA
PCA
PCA
PCA
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Integrated Ethernet switch and two (2) RJ-45
Yes
connector interface ports for 10/100 Mbps
connectivity
Integrated PBX System Directory Access
PCA
W/Dial by Name
No
Bluetooth interface for wireless headset
Yes
USB Interface
Yes
Power over Ethernet
Gigabit (10/100/1000 Mbps) Ethernet
Available in May/June
connectivity
2007 (Expected)
Available now
Embedded Web-browser applications
A picture of this terminal is included in the attached Dterm PowerPoint file.
4.2.4 Executive Desktop IP Telephone Instrument
The Professional model will be used by VoiceCon’s executive management
team. It should have, at minimum the following design attributes and
features/functions:
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
12 key dial pad
Twelve (12) programmable line/feature keys with soft label/ status
indicators
G711, G729 and wideband voice codecs
Auto Self Discovery/DHCP
Echo Canceller
QoS Support (802.1p/Q, DiffServ)
Hold key
Last Number Redial key
Release key
Message Waiting/Call Ringing indicator(s)
Full Duplex Speakerphone
Speaker/Mute key
Volume Control keys/slide
High resolution, backlit, color pixel-based, graphical display screen
with four (4) associated context sensitive soft feature lablels (key,
cursor, or navigator control)
LDAP access
Stored Call Data (Last 10 numbers dialed/Last 10 incoming call
numbers)
Integrated Ethernet switch and two (2) RJ-45 connector interface
ports; 10/100 Mbps connectivity
Headset interface (Bluetooth is also acceptable)
IEEE 802.af POE support
The Professional model must also be capable of supporting the following
integrated feature/functions if required at some future time:
•
Gigabit (10/100/1000 Mbps) Ethernet connectivity
Page 47
VoiceCon Spring 2007
•
Request for Proposal
for an IP Telephony System
Embedded Web-browser applications
Vendor Response Requirement:
Confirm that your proposed Executive model satisfies the stated
requirements and provide a brief product description that includes an
illustration or photograph (PPT format, only) of the instrument. Indicate in
your response any and all requirements not satisfied. State which required
feature-specific keys are not available, but softkey feature access can be
used as an alternative.
NEC Response: NEC has chosen the Dterm IP 320C INASET telephone set to meet
this requirement.
The INASET 320C offers outstanding functionality, intuitive user interfaces, crystal clear
audio quality, stylish design, network manageability and state-of-the-art features. The
INASET was designed for companies with converged IP infrastructures that need to
deliver productivity-enhancing applications and services directly to the desktop. An
enterprise-class IP terminal, the INASET 320C has 4” LCD Color display and 16
programmable keys and softkeys for access to applications such as unified messaging,
company directories and call logs. The applications can be custom-designed or are
provided by the NEC Openwork® package.
Key Features
The INASET line supports a comprehensive suite of advanced applications for mobility
and collaboration and is especially well-suited for retail, hospitality, education and
government. The INASET 320C is one in a series of NEC IP terminals designed to meet
the diverse communication needs of users across the enterprise - from affordable entrylevel IP phones to sophisticated network appliances. All are designed with ergonomics
and modern office aesthetics in mind.
The NEC INASET 320 C is designed with conveniently placed, easy-touse features.
Pixel-based display—A Color 320 x 160 pixel display provides intuitive access to
calling features.
Four softkeys access various services through an external XML server. Here are
some of the basic services provided with the integration of an OpenWorX server
Software Development Kit allows the development of third-party XML-based
applications using the higher level ASP/.NET wrapper or the native INASET XML
specification.
Built-in speaker allows for hands-free operation.
Full duplex audio support when the INASET is set for hands-free mode.
Microphone port provides support for the external NEC microphone to be used in
a hands-free mode.
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VoiceCon Spring 2007
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for an IP Telephony System
Built-in 3-port switch provides automatic negotiation for 10/100Mb network
connection.
16 programmable multi-function keys can be set to access any of the hundreds
of terminal features within the NEAX PBX.
Six function keys fixed to the base of the terminal: Hold, Directory, Message,
Speaker, Microphone and Transfer.
Support for 7Khz of audio when connected to another1 INASET product.
Support for a multitude of powering options: Local power, Ethernet power via
802.3af or Cisco Discover Protocol. Also supports the NEC discovery protocol
with the ILPA dongle integration.
The INASET 320C also provides an expansion port that allows an
additional feature to be driven at the desktop:
With the integration of the AD (A) adapter, you can now equip your desktop
terminal for external recording capabilities. Incoming and outgoing calls can be
recorded to a local PC hard drive or tape recorder which is connected to the
INASET 320G.
Or add the PS (A) adapter to the INASET 320G which provides survivability to
the desktop. If the IP network crashes, your INASET has a backup PSTN circuit
that provides incoming and outgoing call capabilities. This is a perfect add-on for
users who require INASET power but are located miles or even hundreds of
miles away from their servicing switch.
Terminal Attribute (Dterm IP 320C INASET)
12 key dial pad
Sixteen (16) programmable line/feature keys
with soft label/status indicators
Voice Codecs:
G.711
G.729A
Wideband
Auto Self Discovery/DHCP
Echo Canceler
QoS Support
802.1p/Q
DiffServ
Hold Key
Last Number Redial Key
Release Key
Message Waiting / Call Ringing Indicator
Requirement
Supported
Yes
Yes, shown 8 at a time
on the display
Yes
Yes
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Page 49
VoiceCon Spring 2007
Two-Way Full Duplex Speakerphone
Speaker/Mute Key
Volume Control Keys/Slide
High resolution, backlit, Color pixel-based,
graphical display screen with four (4)
associated context sensitive soft feature
labels (key, cursor, or navigator control)
Request for Proposal
for an IP Telephony System
Yes
Yes
Yes
Yes
PCA
LDAP Acess
PCA
Stored Call Data
PCA
Last Incoming Calls (10, Minimum)
PCA
Last Numbers Dialed (10, minimum
Integrated Ethernet switch and two (2) RJ-45
Yes
connector interface ports for 10/100 Mbps
connectivity
Integrated PBX System Directory Access
PCA
W/Dial by Name
No
Bluetooth interface for wireless headset
Yes
USB Interface
Yes
Power over Ethernet
Gigabit (10/100/1000 Mbps) Ethernet
Available in May/June
connectivity
2007 (Expected)
Available now
Embedded Web-browser applications
A picture of this terminal is included in the attached Dterm PowerPoint file.
4.2.5 Desktop IP Telephone Instrument Web-browser Functionality
Vendor Response Requirement:
Provide a brief description of embedded Web-browser functionality for the
proposed Professional and Executive IP desktop telephone instrument
models. Include the following information in your response: browser protocol
(HTML, XML, WAP, Java, LDAP, Stimulus, other); station user interaction
(touchscreen and/or keypad control cursor control; ability to place calls
during active screen applications; screen saver option; standard and optional
applications (visual mailbox; personal directory and calendar; web page
access and display; visual alerts; audio alerts; et al).
NEC Response: Among the high value business applications created by NEC for the
INASET are: OpenWorX® directory functionality, call log history for both incoming and
outgoing calls, location status and message reader. NEC also makes available an SDK
(Software Development Kit), which allows 3rd party developers the tools necessary to
code their own telephony service for the INASET product line.
Software Development Kit allows the development of third-party XML-based
applications using the higher level ASP/.NET wrapper or the native INASET XML
specification.
Page 50
VoiceCon Spring 2007
4.2.5
Request for Proposal
for an IP Telephony System
Desktop Instrument Options and Add-on Modules
Vendor Response Requirement:
Provide a brief description of all hardware/software options and/or add-on
modules currently available with the proposed Economy, Administrative,
Professional, and Executive models. Options/modules should include key
modules, display modules, BlueTooth interface, USB interface, Gigabit
Ethernet connectors, et al. necessary to satisfy the above telephone model
requirements. Indicate the specific models that support the individual
option/module.
NEC Response: The following adapters are available on all four of the above model
telephones:
With the integration of the AD (A) adapter, you can now equip your desktop
terminal for external recording capabilities. Incoming and outgoing calls can be
recorded to a local PC hard drive or tape recorder which is connected to the
Dterm.
Or add the PS (A) adapter which provides survivability to the desktop. If the IP
network crashes, your Dterm has a backup PSTN circuit that provides incoming
and outgoing call capabilities. This is a perfect add-on for users who require
Dterm power but are located miles or even hundreds of miles away from their
servicing switch. Note that a dedicated PSTN line is required for this adapter.
4.2.6
SIP Compatibility
It is desirable, but not required, that the proposed desktop IP telephone
instruments conform to current SIP standards and specifications at time of
installation and system cutover. If any or all the proposed instrument models
do not support natively embedded SIP capabilities as proposed, then it is
acceptable that a firmware download upgrade be available when requested.
Vendor Response Requirement
Indicate which of the proposed telephone models are native SIP or can be
reprogrammed via a SIP firmware download when requested by VoiceCon.
Identify which proposed models cannot currently be reprogrammed for SIP
support at this time.
NEC Response: Only the Dterm IP ITR-4D can be reprogrammed into a SIP compatible
terminal. The INASETS cannot be converted to SIP.
4.3
PC Client Softphone
A PC client softphone will be used by station users and attendant operators
as their primary desktop voice terminal. It is desirable, but not mandatory,
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VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
that the PC client softphone application conform to SIP standards and
specifications.
Vendor Response Requirement
Indicate if the proposed softphone solution satisfies the stated SIP
requirement. If the proposed softphone solution does not support SIP, does
your product portfolio currently include a PC client softphone solution that
does?
NEC Response: The Dterm SP30 allows VoiceCon users to capitalize on the
advantages of a converged voice and data network whether they’re in the office or on
the road. The Dterm SP30 combines traditional business communication needs with the
data applications VoiceCon requires.
Designed to meet diverse customer needs as a primary desktop telephone, a
supplemental desktop telephone or a telecommuting device, the Dterm SP30 software
phone provides an easy means of increasing interpersonal communications through the
NEAX® backbone.
The Dterm SP30 optimally delivers high quality voice via a USB-connected headset and
handset or via a PC sound card, microphone and speakers. With a simple drag and
drop, the Dterm SP30 allows telephone dialing from other telephone directory
applications such as Microsoft Outlook®, HTML pages, Microsoft Word documents, etc.
In addition, the Dterm SP30 provides an interface to Microsoft’s Telephony Application
Programming Interface (TAPI), allowing TAPI-enabled applications, such as Outlook and
ACT!®, to make and receive calls.
The Dterm SP30 can be displayed in 1 of 5 different GUIs. The Dterm SP30 also allows
for 3 different modes of operation:
Maximized mode: Access to full line of SoftPhone features such as application
sharing, member lists, conference mode, chatting capabilities, Internet access
and many others are just one click away.
Compact Mode: This mode is an L-shaped user interface, operating in a small
footprint on the PC screen. Compact view allows the SoftPhone to remain active
while another application window such as a Word document, database file or
email is the primary focus on the PC. With the compact view, the most popular
features of the converged SoftPhone are just a click away.
Task Mode: The SoftPhone can be minimized and shown as a task within a
Microsoft Operating system. While operating in this mode, the SoftPhone will
output an audio notification to the user upon receiving an incoming call. It will be
up to the user to utilize the hot key in order to activate the Dterm SP30
application and answer the call. The Dterm SP30 has 32 Line/Function keys for
features and line appearances.
A picture of this terminal is included in the attached Dterm PowerPoint file.
4.3.1
Desktop Station User Application
The proposed PC client softphone solution must be able to support a
minimum of six programmable line appearances, integrated system and
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VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
personal directories with search/dial-by-name capabilities, and functions
comparable to the proposed Professional model. The softphone solution
must also be able to support a peripheral headset.
Vendor Response Requirement
Confirm that the proposed softphone solution satisfies the stated
requirements and provide a brief product description that includes an
illustration/photograph (PPT format, only) that depicts the look and feel of an
active call screen display.
NEC Response: The Dterm SP30 allows VoiceCon users to capitalize on the
advantages of a converged voice and data network whether they’re in the office or on
the road. The Dterm SP30 combines traditional business communication needs with the
data applications VoiceCon requires.
Designed to meet diverse customer needs as a primary desktop telephone, a
supplemental desktop telephone or a telecommuting device, the Dterm SP30 software
phone provides an easy means of increasing interpersonal communications through the
NEAX® backbone.
The Dterm SP30 optimally delivers high quality voice via a USB-connected headset and
handset or via a PC sound card, microphone and speakers. With a simple drag and
drop, the Dterm SP30 allows telephone dialing from other telephone directory
applications such as Microsoft Outlook®, HTML pages, Microsoft Word documents, etc.
In addition, the Dterm SP30 provides an interface to Microsoft’s Telephony Application
Programming Interface (TAPI), allowing TAPI-enabled applications, such as Outlook and
ACT!®, to make and receive calls.
The Dterm SP30 can be displayed in 1 of 5 different GUIs. The Dterm SP30 also allows
for 3 different modes of operation:
Maximized mode: Access to full line of SoftPhone features such as application
sharing, member lists, conference mode, chatting capabilities, Internet access
and many others are just one click away.
Compact Mode: This mode is an L-shaped user interface, operating in a small
footprint on the PC screen. Compact view allows the SoftPhone to remain active
while another application window such as a Word document, database file or
email is the primary focus on the PC. With the compact view, the most popular
features of the converged SoftPhone are just a click away.
Task Mode: The SoftPhone can be minimized and shown as a task within a
Microsoft Operating system. While operating in this mode, the SoftPhone will
output an audio notification to the user upon receiving an incoming call. It will be
up to the user to utilize the hot key in order to activate the Dterm SP30
application and answer the call. The Dterm SP30 has 32 Line/Function keys for
features and line appearances.
4.3.1.1 Teleworker Station User Application
The proposed PC client softphone solution may also be used by some station
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VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
users as a teleworker voice terminal outside the VoiceCon facility
environment.
Vendor Response Requirement
Confirm that the proposed customer premises softphone solution can be used
as an off-premises teleworker voice terminal option . Indicate in your
response if any optional hardware and/or software requirements are required
to support teleworker mode operations for deployment in a home, hotel, or
office environment.
NEC Response: The Dterm SP30 allows VoiceCon users to capitalize on the
advantages of a converged voice and data network whether they’re in the office or on
the road. The Dterm SP30 combines traditional business communication needs with the
data applications VoiceCon requires.
Designed to meet diverse customer needs as a primary desktop telephone, a
supplemental desktop telephone or a telecommuting device, the Dterm SP30 software
phone provides an easy means of increasing interpersonal communications through the
NEAX® backbone.
The Dterm SP30 optimally delivers high quality voice via a USB-connected headset and
handset or via a PC sound card, microphone and speakers. With a simple drag and
drop, the Dterm SP30 allows telephone dialing from other telephone directory
applications such as Microsoft Outlook®, HTML pages, Microsoft Word documents, etc.
In addition, the Dterm SP30 provides an interface to Microsoft’s Telephony Application
Programming Interface (TAPI), allowing TAPI-enabled applications, such as Outlook and
ACT!®, to make and receive calls.
The Dterm SP30 can be displayed in 1 of 5 different GUIs. The Dterm SP30 also allows
for 3 different modes of operation:
Maximized mode: Access to full line of SoftPhone features such as application
sharing, member lists, conference mode, chatting capabilities, Internet access
and many others are just one click away.
Compact Mode: This mode is an L-shaped user interface, operating in a small
footprint on the PC screen. Compact view allows the SoftPhone to remain active
while another application window such as a Word document, database file or
email is the primary focus on the PC. With the compact view, the most popular
features of the converged SoftPhone are just a click away.
Task Mode: The SoftPhone can be minimized and shown as a task within a
Microsoft Operating system. While operating in this mode, the SoftPhone will
output an audio notification to the user upon receiving an incoming call. It will be
up to the user to utilize the hot key in order to activate the Dterm SP30
application and answer the call. The Dterm SP30 has 32 Line/Function keys for
features and line appearances.
4.3.2 Soft Attendant Console
Attendant operator console requirements are to be satisfied using a PC client
Page 54
VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
softphone application. The attendant console application should include
several distinct display fields, such as: incoming call queue and active caller
information; release loop keys; feature/function keys; direct station selection
(contact directory)/ busy lamp field; trunk groups; minor/major alarms; and
messaging. GUI capabilities must support drag & click operations.
At minimum the following information and data must be available in the
softphone screen display: # Calls in queue; Call appearance status;
Calling/called party number/name; Trunk ID; COS/COR; # Calls waiting; call
coverage status; time/date, call duration; text messages; alarm notification
Vendor Response Requirement
Confirm the proposed softphone solution satisfies the stated requirements,
and provide a brief description of the proposed softphone solution when
programmed for attendant console operation. Include in the response a
representative illustration or photograph (PPT format, only) that conveys the
look and feel of an active call console display screen.
NEC Response: The Business Attendant System (BAS) is a call-processing and callmanagement application that provides advanced attendant features on a desktop PC.
The application utilizes a friendly Microsoft Windows interface and can be controlled
by a mouse for simplicity or a keyboard for speed. One of the key advantages of the
Business Attendant System is that it is scalable in size, so it can affordably be
implemented by large or small system users. Another key advantage of the BAS is that
the in use licensing provides the flexibility in scheduling operations during the workday.
The application extends the attendant’s abilities to cover calls with database-driven
loops and console features that provide the most effective and efficient call handling and
directory assistance right from the attendant workstation.
An expanded directory brings critical status information as well as one-touch dialing or
forwarding capabilities into one comprehensive multifunctional directory window. It’s
easy to read and easy to use, so training is a breeze. Database information for the
Business Attendant System is stored in the same general database utilized by other
OpenWorX applications, such as the Dialer and the Location Status Information. Queries
and modifications to the database are processed dynamically in real time. As a result,
any changes made to the directory data across OpenWorX applications are immediately
reflected in the Business Attendant System, ensuring that everyone in the organization
always has access to the latest information.
The Business Attendant System also provides intelligent message taking capabilities to
ensure that associates receive all of their calls. Messages can be saved for later access
as well as e-mailed to predetermined and new locations. Plus, a lit up message-waiting
lamp indicates the users to the presence of awaiting messages which can be read on
their Dterm.
The Monitored Speed Dials window in the Business Attendant System provides similar
functionality to the Direct Station Selector/Busy Lamp Field. Up to 40 speed dial buttons
are displayed in the window. For setting up the speed dial buttons, simply drag and drop
extensions from the directory window. The current status of a Dterm programmed as a
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VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
speed dial button is reflected on the button itself. When the button is highlighted in Red,
it signifies that the extension is busy or off-hook. By hovering the mouse over the
button, the BAS attendant can see additional information on that extension. Such
additional information could be, whether or not the DND is set, Forwarding is set, and
persons Status. Calls can be transferred to extensions on the speed dial buttons by
simply clicking the appropriate button.
The Business Attendant System may be “Distributed” on the customer’s network,
meaning that the attendants may be scattered across the Local Area Network when
connected directly to a PBX or across a Wide Area Network in an NEC Fusion or IP
Environment. Network design must provide fast response times, as these are real-time
functions on the Attendant’s PC.
A picture of this terminal is included in the attached Dterm PowerPoint file.
4.4
IP Audio Conferencing Unit
VoiceCon requires a limited number of desktop audio conferencing units with
multidirectional, full duplex speakerphone operation. The unit must be
native IP.
Vendor Response Requirement
Provide a brief description of the proposed IP audio conferencing unit and
include in the response an illustration or photograph (PPT format, only) of
the unit.
NEC Response:
NEC Unified Solutions, Inc. has teamed up with ClearOne
Communications and is proposing the Conference Max audio conferencing unit for this
response. All Conference Max models include new tabletop conferencing features.
These NEC labeled units are designed with the same color scheme as the Dterm®
Series i sets, resulting in a stylish addition to the NEC terminal line up.
Corded
Conference Max
Conference Max
Expansion
75
00
750156
Cordless
Conference Max Plus
750074
Page 56
VoiceCon Spring 2007
73
An
alo
g
Wi
red
Request for Proposal
for an IP Telephony System
Analog Wired
•
•
•
•
•
•
Analog Cordless 2.4
GHz
150’ Wireless Range
12 Hours Talk Time
36 Hours Stand-by
Distributed Echo Cancellation effectively eliminates echo
Noise cancellation removes background noises from fans or
HVAC systems
Full-duplex sound enables participants to speak and listen at the
same time without cutting in and out
Automatic level controls keep participants’ audio balanced and
consistent
First-mic priority eliminates hollow “tunnel” sound by activating
only the microphone closest to the person speaking
Three microphones provide 360° audio pickup
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VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
Corded
Conference Max
Conference Max
Expansion
Includes:
1 Max Phone
Pod
Phone Pod Base
25’ Cat 5 cable
Telephone Cable
Documentation
CD
Quick Start
Guide
Includes:
1 Max Phone Pod
12’ Cat 5 cable
Telephone Cable
Documentation CD
Quick Start Guide
•
•
•
•
Small
Conference
Room Up to 8
People
Executive
Offices
Home Offices
* Requires Conf
Max
•
•
•
Medium to large
conference
rooms
Training rooms
Unique Room
Configurations
(U-shaped)
Up to 3
expansion pods
may be added to
the base pod.
Cordless
Conference Max Plus
Includes:
1 Max Phone Plus Pod
Base unit
Battery pack Power supply/charger
Power supply retainer
Telephone cable
Documentation CD
Quick start guide
•
•
•
•
•
Rooms without Phone Lines 150’
from base unit
Small Conference Room Up to 8
People
Executive Offices
Home Offices
A picture of this terminal is included in the attached Dterm PowerPoint file.
Page 58
VoiceCon Spring 2007
4.8
Request for Proposal
for an IP Telephony System
Other IP Telephone Instruments
Please provide a brief description of additional IP desktop telephone
instrument models included in your portfolio other than the models used to
satisfy the Economy, Administrative, Professional, and Executive
requirements. Information should include, at minimum, fixed
feature/function, number of programmable line/feature keys, display
description (if applicable), type of speakerphone (if appplicable), and any
other information you deem vital. Include an illustration/photograph (PPT
format, only) for each of these additional models.
NEC Response: The Dterm IP terminal family expands its capabilities with 3 IP
terminals (8, 16 and 32-button terminals) and 16LD-3 (Desi-less IP terminal).
Enhanced features with Dterm IP Version 3 terminals:
Support for 802.3af power over Ethernet (POE)
There is now support for both the industry standard way of
powering across the Ethernet, 802.3af, and Cisco
Discovery Protocol.
Support for optional adapters to allow for expanded capability
(unique in the Industry)
All Dterm IP Version 3 terminals provide an expansion port
that allows an additional feature to be driven at the
desktop:
ƒ
With the integration of the AD (A) adapter, you can now equip
your desktop terminal for external recording capabilities. Incoming
and outgoing calls can be recorded to a local PC hard drive or
tape recorder that is connected to the Dterm IP Version 3 terminal.
ƒ
Adding the PS (A) adapter to the Version 3 terminal provides
survivability to the desktop. If the IP network crashes, your
INASET® has a backup PSTN circuit that provides incoming and
outgoing call capabilities. This is a perfect add-on for users who
are located miles or even hundreds of miles away from their
servicing switch.
IP terminal offering in a 32-button form
The Dterm IP Version 3 now lets you choose from 8, 16 or
32-button sizes.
•
16 button Desi-less terminal
Call Handling
Page 59
VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
ƒ
Four Local Soft Key Controls (detail functions are dependent on
PBX)
ƒ
Message Waiting LED
ƒ
24 Character, 3-Line LCD
ƒ
Built-in Headset Jack Connector
ƒ
Speed Dial/DSS Buttons
ƒ
Programmable Line Keys to support almost every feature/function
that the TDM Dterm Series i terminal supports
ƒ
11 Dedicated Function keys (Feature, Recall, Conf, Redial, Hold,
Transfer, Answer & Speaker)
Convergence
ƒ
Two 10/100 full duplex Ethernet ports - One connects the Dterm
IP to the local Ethernet network; the other provides connectivity for
a local workstation.
ƒ
Three types of powering options
ƒ
Local AC adapter (optional hardware)
ƒ
In-line power support utilizing the 802.3af and CDP when
connected to Cisco®-based equipment.
ƒ
Transportable QoS follows the user regardless of log-in location.
ƒ
Multiple Voice Coding support automatically negotiates to a
common setting.
ƒ
G.711 provides an international standard for encoding/decoding
telephony signals on a 64 Kbps non-compressed channel.
ƒ
It also supports the compression algorithms G.729A (8Kbps) and
G.723.1 (5.3/6.3 Kbps).
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Request for Proposal
for an IP Telephony System
5.0 Call Processing Features
The proposed communications system should have a robust list of call
processing features supporting station user, attendant, and system
operations.
5.1
Station User Features
It is required that the proposed communications system support the following
list of station user features. Definitions for most listed features may be
found in PBX Systems for IP Telephony (2002), written by Allan Sulkin
and published by McGraw-Hill Professional.
Table 9 Station User Features
STATION USER FEATURES
ADD-ON CONFERENCE (6 party or more)
AUTOMATIC CALLBACK
AUTOMATIC INTERCOM
BRIDGED CALL APPEARANCE
CALLBACK LAST INTERNAL CALLER
CALL COVERAGE (PROGRAMMED)
INTERNAL & EXTERNAL CALL PROGRAMMING
TIME OF DAY/DAY OF WEEK CALL PROGRAMMING
ANI/DNIS/CLID CALL PROGRAMMING
INTERNAL CALLER ID PROGRAMMING
CALL FORWARDING - ALL CALLS
CALL FORWARDING - BUSY/DON'T ANSWER
CALL FORWARDING - FOLLOW-ME
CALL FORWARDING - OFF-PREMISES
CALL FORWARDING: RINGING
CALL HOLD
CALL PARK
CALL PICKUP - INDIVIDUAL
CALL PICKUP - GROUP
CALL TRANSFER
CALL WAITING
CONSECUTIVE SPEED DIALING
CONSULTATION HOLD
CUSTOMER STATION REARRANGEMENT
DIAL BY NAME
DISCRETE CALL OBSERVING
DISTINCTIVE RINGING
DO NOT DISTURB
ELAPSED CALL TIMER
EMERGENCY ACCESS TO ATTENDANT
EXECUTIVE ACCESS OVERRIDE
EXECUTIVE BUSY OVERRIDE
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VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
FACILITY BUSY INDICATION
GROUP LISTENING
HANDS-FREE DIALING
HANDS-FREE ANSWER INTERCOM
HELP INFORMATION ACCESS
HOT LINE
INCOMING CALL DISPLAY
INDIVIDUAL ATTENDANT ACCESS
INTERCOM DIAL
LAST NUMBER REDIALED
LINE LOCKOUT
LOUDSPEAKER PAGING ACCESS
MALICIOUS CALL TRACE
MANUAL INTERCOM
MANUAL ORIGINATING LINE SERVICE
MEET ME CONFERENCING (6-Party or more)
MESSAGE WAITING ACTIVATION
MULTI-PARTY ASSISTED CONFERENCE w/SELECTIVE CALL DROP
MUSIC ON HOLD
OFF-HOOK ALARM
PADLOCK
PAGING/CODE CALL ACCESS
PERSONAL CO LINE (PRIVATE LINE)
PERSONAL SPEED DIALING
PERSONALIZED RINGING
PRIORITY CALLING
PRIVACY - ATTENDANT LOCKOUT
PRIVACY - MANUAL EXCLUSION
RECALL SIGNALING
RINGER CUT-OFF
RINGING TONE CONTROL
SAVE AND REDIAL
SECONDARY EXTENSION FEATURE ACTIVATION
SEND ALL CALLS
SILENT MONITORING
STEP CALL
STORE/REDIAL
SUPERVISOR/ASSISTANT CALLING
SUPERVISOR/ASSISTANT SPEED DIAL
TEXT MESSAGES
TIMED QUEUE
TRUNK FLASH
TRUNK-TO-TRUNK CONNECTIONS
WHISPER PAGE
Vendor Response Requirement
Confirm that the proposed communications system supports each of the
above listed station user features. Identify any and all features that are not
included as part of the standard call processing software generic package.
Identify any and all of the listed features that require additional hardware
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VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
and/or software, e.g., CTI application server, because they are not included
as part of the standard generic software package.
NEC Response: A complete list of features is attached in the Appendices section of this
response. Only the Call Conferencing and Paging Features require extra equipment.
5.1.1
Additional Station User Features
Vendor Response Requirement
Provide a listing of proposed standard generic software station user features
that are not included in Table 9 that VoiceCon may find of use and benefit.
NEC Response: A complete list of features is attached in the Appendices section of this
response.
5.2
Attendant Operator Features
It is required that the proposed communications system support the following
list of attendant operator features. Definitions for most listed features may
be found in PBX Systems for IP Telephony (2002), written by Allan Sulkin
and published by McGraw-Hill Professional.
Table 10 Attendant Operator Features
ATTENDANT OPERATOR FEATURES
AUTO-MANUAL SPLITTING
AUTO-START/DON'T SPLIT
BACK-UP ALERTING
BUSY VERIFICATION OF TERMINALS/TRUNKS
CALL WAITING
CAMP-ON
CONFERENCE
CONTROL OF TRUNK GROUP ACCESS
DELAY ANNOUNCEMENT
DIRECT STATION SELECTION w/BLF
DIRECT TRUNK GROUP SELECTION
DISPLAY
INTERCEPT TREATMENT
INTERPOSITION CALL & TRANSFER
INTRUSION (BARGE-IN)
OVERFLOW
OVERRIDE OF DIVERSION FEATURES
PAGING/CODE CALL ACCESS
PRIORITY QUEUE
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VoiceCon Spring 2007
Request for Proposal
for an IP Telephony System
RECALL
RELEASE LOOP OPERATION
SERIAL OPERATION
STRAIGHT FORWARD OUTWARD COMPLETION
THROUGH DIALING
TRUNK-TO-TRUNK TRANSFER
TRUNK GROUP BUSY/WARNING INDICATOR
TRUNK ID
Vendor Response Requirement
Confirm that the proposed communications system supports each of the
above listed attendant operator features. Identify any and all features that
are not included as part of the proposed standard generic software feature
package. Identify any and all features that require additional hardware
and/or software, e.g., CTI application server, not standard with the proposed
system model(s).
NEC Response: A complete list of features is attached in the Appendices section of this
response.
5.2.1
Additional Attendant Operator Features
Vendor Response Requirement
Provide a listing of proposed standard generic software attendant operator
features that are not included in Table 10 that VoiceCon may find of use and
benefit.
NEC Response: A complete list of features is attached in the Appendices section of this
response.
5.3
System Features
It is required that the proposed communications system support the following
list of system features. Definitions for most listed features may be found in
PBX Systems for IP Telephony (2002), written by Allan Sulkin and
published by McGraw-Hill Professional.
Table 11 System Features
SYSTEM FEATURES
ACCOUNT CODES
ADMINISTERED CONNECTIONS
ANSWER DETECTION
AUTHORIZATION CODES
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AUTOMATED ATTTENDANT
AUTOMATIC CALL DISTRIBUTION
AUTOMATIC ALTERNATE ROUTING
AUTOMATIC CAMP-ON
AUTOMATIC CIRCUIT ASSURANCE
AUTOMATIC NUMBER ID
AUTOMATIC RECALL
AUTOMATIC ROUTE SELECTION - BASIC
AUTOMATIC TRANSMISSION MEASUREMENT SYSTEM
CALL-BY-CALL SERVICE SELECTION
CALL DETAIL RECORDING
CALL LOG
CENTRALIZED ATTENDANT SERVICE
CLASSES OF RESTRICTION (SPECIFY #)
CLASSES OF SERVICE (SPECIFY #)
CODE CALLING ACCESS
CONTROLLED PRIVATE CALLS
DELAYED RINGING
DIAL PLAN
DIALED NUMBER ID SERVICE
DIRECT DEPARTMENT CALLING
DIRECT INWARD DIALING
DID CALL WAITING
DIRECT INWARD SYSTEM ACCESS
DIRECT INWARD TERMINATION
DIRECT OUTWARD DIALING
E-911 SERVICE SUPPORT
EXTENDED TRUNK ACCESS
FACILITY RESTRICTION LEVELS
FACILITY TEST CALLS
FIND ME- FOLLOW ME
FORCED ENTRY ACCOUNT CODES
HOTELING (/PERSONAL ROAMING)
HOUSE PHONE
HUNTING
INTEGRATED SYSTEM DIRECTORY
LEAST COST ROUTING (Tariff-based, TOD/DOW)
MULTIPLE LISTED DIRECTORY NUMBERS
MUSIC ON HOLD
NIGHT SERVICE –FIXED
NIGHT SERVICE - PROGRAMMABLE
OFF-HOOK ALARM
OFF-PREMISES STATION (OPX)
OPEN SYSTEM SPEED DIAL
PASSWORD AGING
POWER FAILURE TRANSFER STATION
RECENT CHANGE HISTORY
RESTRICTION FEATURES
CONTROLLED
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FULLY RESTRICTED
INWARD/OUTWARD
MISCELLANEOUS TERMINAL
MISCELLANEOUS TRUNK
TOLL/CODE
TRUNK
VOICE TERMINAL (IN/OUT)
ROUTE ADVANCE
SECURITY VIOLATION NOTIFICATION
SHARED TENANT SERVICE
SNMP SUPPORT
SYSTEM SPEED DIAL
SYSTEM STATUS REPORT
TIME OF DAY ROUTING
TIMED REMINDER
TRUNK ANSWER ANY STATION
TRUNK CALLBACK QUEUING
UNIFORM CALL DISTRIBUTION
UNIFORM DIAL PLAN
VIRTUAL EXTENSION
VOICE MESSAGE SYSTEM INTERFACE
5.3.1 Additional System Features
Vendor Response Requirement
Provide a listing of proposed standard generic software system features that
are not included in Table 11 that VoiceCon may find of use and benefit.
NEC Response: A complete list of features is attached in the Appendices
section of this response.
5.4 Mobility Features
VoiceCon requires that the proposed IPTS support a variety of features and
applications to support its mobile workforce.
5.4.1
Fixed Teleworking
VoiceCon requires that its employees be able to use a PC client softphone
outside the office environment using Internet or VPN access.
Vendor Response Requirement
Verify that a VoiceCon employee can use a PC client softphone to access the
full set of HQ IPTS features and functions from a remote location using
Internet or VPN access. Specify if there are known NAT or firewall
transversal issues with this application.
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NEC Response: Confirmed, a VoiceCon employee can use the SP-30 Softphone on a
PC at home or at a remote site with a VPN connection to the HQ LAN to access the full
range of features and functions available “in the office.”
5.4.2
Cellular Extensions
VoiceCon requires that its employees be able to utilize their cellular handsets
to answer and place calls that are routed through the HQ IPTS.
Vendor Response Requirement
Verify that the proposed IPTS solution can support cellular handsets as
system extensions, and briefly describe the IPTS option. Off-premises call
forwarding of calls directed to an IPTS extension is not satisfactory for this
requirement. Specifically address the following:
5.4.2.1
Shared directory number with IP desktop telephone instrument or
PC client softphone for inbound and outbound calls
NEC Response: The MobileConnex solution allows for a high level of
accessibility because your office number is bridged to your mobile phone.
Since both phones ring simultaneously, you have the option of answering the
call on your office IP Dterm set, SP30 Softphone or on your remote phone
With the MobileConnex application and based on COS (class of service),
mobile workers are now able to answer incoming calls from wherever they
are, transfer calls, manage conference calls and dial internal 4- or 5-digit
extensions directly from their cellular, mobile or home phones.
5.4.2.2
Access to IPTS call answering and calling features (identify specific
features available to cellular handset user)
NEC Response: With the MobileConneX solution, mobile users can access
PBX dialtone and features. Any cellular telephone connected to the
MobileConneX application is able to leverage corporate PBX/KTS features
like internal 4 or 5-digit dialing, hold, conference and transfer, from anywhere.
The MobileConneX application is capable of translating dial pad key presses
generated by cellular phones into PBX/KTS specific digital signaling. The
converted signals sent from the MobileConneX application to a NEC PBX
look like signals produced by pressing certain keys on a feature-rich digital
handset.
5.4.2.3
Shared voice mailbox for desktop stations and cellular extensions
NEC Response: The MobileConnex solution allows for a high level of
accessibility because your office number is bridged to your mobile phone.
Since both phones ring simultaneously, you have the option of answering the
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call on your office Dterm set, SP30 Softphone or on your remote phone. The
ring no answer of the “Dterm” determine both the time that the call will hear
ring no answer and all voice mail messages will be stored in the Dterm
station’s voice mailbox.
5.4.2.4
If available provide a picture or photograph in PowerPoint format
of an optional graphical user interface screenshot for use with a
cellular handset
NEC Response: Not Available.
5.4.2.5
Indicate if the cellular extension feature is proprietary to a specific
cellular carrier operator service
NEC Response: The MobileConnex solution provides one-number portability
that is cellular standard independent. All cellular standards are supported
(TDMA, Time Division Multiple Access, CDMA: Code Division Multiple
Access and GSM: Global System for Mobile Communications).
5.4.2.6
Identify any optional hardware/software required to support the
cellular extension feature if not a proposed IPTS generic software
feature
NEC Response: The MobileConnex application requires at least one IP
Dterm Gateway (dependent on number of users) and dual-mode ports on the
digital station card.
5.4.3
Fixed Mobile Convergence
VoiceCon may be interested in implementing a Fixed Mobile Convergence
(FMC) solution at some future date to increase station user productivity and
performance. FMC supports seamless communications between a premises
WLAN and a service provider cellular network using the same mobile
communications device, with access to and implementation of IPTS features
and functions.
Vendor Response Requirement
Briefly describe current efforts and activities to support a FMC solution using
a dual mode 802.11/GSM mobile communications device behind the
proposed IPTS. Include in the response estimated availability dates of the
FMC solution, required WLAN equipment to support premises roaming
capabilities, QoS, and security. Also identify the means to provide seamless
handoff between the 802.11 WLAN and the cellular network for active calls.
NEC Response: NEC Unified Solutions, Inc. is addressing the dual-mode market where
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one single instrument can be used as an in-building wireless handset and a cellular
phone when not in the building.
NEC is currently in the process of introducing Assured Mobility.
UNIVERGE Assured Mobility from NEC
So advanced, it can turn your wireless LAN into a competitive advantage.
UNIVERGE® Assured Mobility™ is NEC’s newest generation wireless LAN (WLAN)
communications solution, connecting people to people and people to the information
they need anytime, anywhere, on any device.
Assured Mobility lets you seamlessly roam on and off campus, from your wireless LAN
to cellular networks and back again with one device. You enjoy true mobility with the
high-quality voice communications you’ve come to expect from NEC.
UNIVERGE Wireless Optimized Architecture™ (WOA) powers Assured Mobility and
reduces network bottlenecks by intelligently routing voice packets directly between
Access Points (APs).
Assured Mobility optimizes your mobility experience while delivering voice, data,
multimedia and many productivity-boosting advanced solutions.
You are assured of the reliability, scalability, investment protection, service and support
you’re accustomed to receiving as part of NEC UNIVERGE solutions.
Deployment is easy, training is fast, and management is centralized and secure. And
NEC’s modularized, rock-solid family of Assured Mobility solutions fully integrates with
other UNIVERGE solutions.
UNIVERGE Wireless Optimized Architecture from NEC
It’s the Power Behind Assured Mobility™
NEC’s UNIVERGE® Wireless Optimized Architecture™ is the unique wireless LAN
(WLAN) that connects people to people and information—anytime, anywhere, on any
device.
Wireless Optimized Architecture (WOA) advances the integration of voice telephony with
WLAN technology to a new level.
NEC’s WLAN takes full advantage of NEC’s unique and advanced Wireless SIP
features, optimizing and enhancing users’ Voice over WLAN (VoWLAN) experience.
In addition, WOA offers enhanced Thin Access Points capabilities with a fully featured
Controlling AP (WL1700-MS), ideal for SMB, pilot, small or remote office deployments.
Along with high-quality voice, WOA offers you e-mail and instant communications, realtime multimedia and data, and many other productivity-boosting advanced solutions.
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WOA also offers advanced and intelligent switching techniques with NEC’s Direct AP-toAP path, which provides peer-to-peer over WLAN.
WOA reduces network traffic congestion by intelligently forwarding media packets
directly between Access Points (APs). This gives you the high-quality voice
communications service you’ve come to expect from NEC.
Call hand-offs are virtually flawless, minimizing dropped calls. And WOA reduces data
bottlenecks at the controller, extending equipment life and delivering superb call quality.
WOA drives all functions of the Assured Mobility Solution delivering best-in-class secure
mobility with a broad range of converged mobility terminals.
WOA is a truly unified mobility platform comprised not only of the new UNIVERGE WL
Controllers and APs, but also one that features a tight integration with advanced
management capabilities. Such capabilities are enabled and made possible using
WOA’s WLMS management software, built to allow for centralized mobility management
as well as on-target planning, deployment and maintenance.
Centralized management is enhanced with WOA. Such enhancements are proven with
the ability of such a platform to manage controllers and APs independent of their location
in the network’s LAN or WAN environment. WOA offers flexibility in deploying an
advanced WLAN, but with the benefit of centralized management.
This centralized management is not limited to the infrastructure components of this
WLAN, but is also extended to users and clients that may roam seamlessly and securely
throughout the network. WOA’s architecture enables WLMS to provide detailed reports
of clients’ profiles, historical usages, and roaming activities; therefore extending the
network’s control beyond just the infrastructure. This enablement and control of clients
and their activities allows the advanced WLAN to be more forgiving for real-time
applications (such as voice) that are greatly affected by events such as multiple login
attempts due being mobile and roaming from one AP to another. WOA is an
enhancement to the WLAN with features specific to voice-over-wireless LAN (VoWLAN).
Thanks to WOA, you’ll enjoy the same network reliability, scalability, investment
protection, service and support that you’re accustomed to receiving as part of all NEC
UNIVERGE solutions.
High-quality Voice Communication. NEC’s switching and handset designs
prioritize voice calls, complete hand-offs quickly and seamlessly, and effectively
control call admission—all critical for true wireless mobility.
Simplified Deployment. The NEC family of WLAN Access Points makes it
simple to deploy and scale the WLAN network as your business grows.
Thin Acess Point with Controller Features. A controlling and intelligent Thin
AP with Access Point and Controller Features provides a low-entry point for small
and medium-size organizations that wish to implement a WLAN.
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Effective Management Tool. A simple-to-use interface helps you design, set up,
operate, and maintain your WLAN.
Reduced Network Bottlenecks. WOA forwards management packets to the
new UNIVERGE WL Controllers (WL1700-MS, WL5050 and WL5100) while
intelligently forwarding media packets directly between Access Points (APs). In
this way, WOA reduces network bottlenecks, ensures high-quality voice service
and extends equipment life.
Uninterrupted Service. Dual-home AP design creates redundancy. Each AP
can have two links into a network, which provides for redundancy and failover
capability. There is no single point-of-control, so reliability is assured without
comprising centralized management. Mission-critical applications are always on.
Investment Protection. The UNIVERGE Assured Mobility WLAN is modular and
scalable with a large portfolio of products, applications and solution sets to meet
diverse needs. As technology advances and your organization grows and
changes, NEC offers flexible licensing for easy upgrades.
Advanced Solutions
Part of the NEC UNIVERGE® Assured Mobility™ promise made possible by a
variety of advanced Communication Solutions.
In addition to true mobility, quality voice, scalability, security and investment protection,
Assured Mobility means your Wireless LAN comes with productivity-boosting, leadingedge communication solutions:
•
Unified Messaging. Combines voice, fax and e-mail messages in one in-box.
Helps your people work smarter, faster.
•
Location Services. WOA enables rapid location lookup of any RFID-tagged
property or person. Provides movement in real-time and historical movement
tracking. Delivers sophisticated enterprise asset management.
•
Presence Information. Tells you where your people are and their availability so
you can make smarter decisions about who to contact, where and when.
•
Intelligent Call Redirection. Redirects voice calls and instant messaging based
on the most up-to-date Presence Information.
•
Wide Variety of Mobility Handsets. Choose from hands-free, dual-mode PDAs,
smart phones, WLAN phones, softphones, wearable devices and more.
•
Customized Mobile Experience. Control your communications from device
choices to application configurations. Provides high-performance features that
are user-friendly and intuitive.
•
Simultaneous Ringing. Advances one-number portability to a new level. When
you place a call, it rings simultaneously on all the called party’s devices. After the
call is established on one device, the user can transfer the call between devices.
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For example, a call is established on a cell phone as a person drives to work.
Then, upon arrival, the person can transfer the call to their office phone.
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6.0.0 Systems Management
The proposed communications system must be administered, monitored, and
maintained through operations organized into five functional areas: Fault,
Configuration, Accounting, Performance and Security. All of the systems and
devices in your proposed solution should attempt to provide comprehensive
operations in each area.
Operations for each area must be accessible through one interface regardless
of the underlying system or device being managed. If a proxy server is used
for intermediate operations, there must be at most one central database for
each functional area. Systems or devices may be accessed individually if no
proxy server is used.
EXCEPTION: Optional call center solutions may provide its own set of FCAPS
management operations separate from the general enterprise
communications solution.
Any supplied management applications must support decentralized access
from any distributed PC client across the HQ LAN/WAN infrastructure and
remote dial-up PC clients. It is also desirable for the applications to support
a browser based user interface for intensive remote operations.
Any supplied management applications may integrate information from the
five functional areas at the presentation level.
Vendor Response Requirement
Confirm and verify that each functional area required to manage the
proposed IPTS network is supported by a single, centrally located proxy
server or, alternatively, each system or device supports a single API
for a given functional area. Provide a brief description of the proposed
management system, including its major hardware and software
components. Specify if the proposed systems management server and
software is available as a bundled offering, only, or if VoiceCon is responsible
for providing its own server hardware to operate the software. If third party
technology is used, please indicate which components are managing your
solution in a vendor agnostic fashion.
NEC Response: The MA4000 Management System is a Web based management
platform for making all types of configuration changes to the Voice Systems themselves.
The MA4000 has the following major components:
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MA4000 Manager - This component is the core of the configuration
management piece. This is a web based, GUI-driven management
platform that is simple to use.
MA4000 Assistant - This component is the end-user interface to *their*
own phone. This interface allows the end-user to change items such as
their call forwarding, speed dials, button programming, etc…
MA4000 Voice Mail Interface - This component is a tightly integrated
interface to the NEC (and other) Voice Mail systems. This allows for
automatic and manual provisioning of the voice mail boxes and features.
MA4000 Traffic Manager - This component generates reports based on
such items as route/trunk traffic and peg count, and processor
occupancy. The user can also define traffic thresholds and notifications
and perform automatic erlang calculations.
MA4000 VoIP Quality Manager - This component generates reports
based on VoIP traffic and QOS statistics. There are also thresholds
definitions to generate alarm notifications for events that exceed defined
parameters.
MA4000 Auth Code Manager - This proactive authorization code
manager is the centralized authentication code engine for the entire Voice
System network. The engine has proactive security configurations
allowing for the shut down of devices attempting to bypass authorization
code security for the purpose of toll fraud.
MA4000 LDAP Auto Provisioning System - This unique component has
the capability of automatically creating new fully configured stations
based on new entries in the customer’s LDAP directory.
MA4000 Installation Manager
The MA4000 Installation Manager is a thin Windows based application that was
designed for the express purpose of installing new Voice Systems. This application has
the capability of importing data from an external source (HR Database, CSV file, etc…)
for the purpose of creating the station data within the Voice System itself. This allows for
a more consistent database installation in the Voice System. The amount of time it now
takes to install and program a new Voice System can be reduced from days/weeks to
hours/minutes using this application. The MA4000 Installation Manager has the
capability of taking a new Voice System from a blank database to a fully configured
system with station, group, trunk, LCR, feature and hardware programming ready to
make and receive calls.
MTS Application Suite (3rd Party Application)
MTS AS provides integrated pro-active and policy based solutions for organizations,
ranging from internal IT operations through to complete IT service management,
including customer care and billing. The MTS application is a web based call accounting
and tele-management application and is fully integrated with the MA4000 Management
System for Single Point of Entry.
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The basic package provided by MTS includes the following modules:
Call Accounting - Collect and monitor IT activity, analyze the costs
involved with the different activities and allocate them to the different
organization units / personnel.
Billback - Expand the Call Accounting capabilities and provide the tools to
charge the users for the different activities (usage or fix)
Budget Control - Monitor pre defined budgets for specific persons /
organization units to prevent unexpected operation costs
Excessive usage control - Alert on unexpected usage patterns detected
by the system
Utilization & Performance Control - Monitor and notify on exceeding
certain pre defined usage rules.
Directory - Easy and user friendly interface for the organization phone
directory.
System Reports - Various range of reports providing analysis and
statistics tools for the system users and managers
All of the above software is offered as either a bundled solution with a
server, or can be sold individually to be installed on a system meeting the
minimum requirements for the software.
6.0.1 System/Port Capacity
Vendor Response Requirement
Identify the maximum number of independent IPTS communications systems
that can be supported by the proposed systems management server, and the
maximum number of user ports that can be passively and actively supported.
NEC Response: No theoretical limit on either. This is based more on processor power
and RAM of the management system server.
6.0.2 Terminal Capacity
Vendor Response Requirement
Identify the maximum number of configurable and active PC client terminals
that can be configured as part of the proposed management server system.
NEC Response: No limit. Processor Licenses for the Microsoft IIS can be purchased.
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6.0.3 Support for Open Standards
The proposed management system should provide support for open
protocols, such as LDAP and SNMP. The proposed management system
should use open encoding schemes, such as XML and HTML.
NEC Response: The Management system is web based and utilizes standards based
on HTML, XML, LDAP and SNMP.
Vendor Response Requirement
Briefly discuss the open standards included in your proposed management
system that supports administration, operations and maintenance services.
Indicate if any protocols or encoding schemes are de facto standards or are
being implemented publicly by other vendors.
NEC Response: LDAP Auto-Provisioning Service is supported, SNMP is supported by
the Voice Servers directly. The NEC MA4000 can use the information stored in the
LDAP directory to automatically provision the Voice Server. Records created in another
application, such as Microsoft Active Directory, can then be retrieved, translated and
applied to the MA4000 provisioning engine and subsequently the Voice Server as well.
SNMP is also used to collect performance and alarm data from
the voice system as well.
.XML is used throughout the product to store configuration
information and for import/export.
.HTML is the core interface for the product and can be displayed
in either secured (SSL) or unsecured mode.
6.0.3 Security Features
Unauthorized access to the communications system is a major concern. The
ability to detect security problems is desirable beyond mechanisms to
prevent security problems.
Vendor Response Requirement
Briefly describe the security features that are embedded in the proposed
management system to prevent unauthorized access and operation. Specify
if media encryption is used for command signaling transmissions. What, if
any, Denial of Service (DoS) and user authentication mechanisms are
supported for the systems management application?
NEC Response: The MA4000 uses a centralized management system called NEC CAS
or centralized authentication service to ensure that only those that are authorized to use
the software actually can.
NEC CAS can use different means of authentication, including SQL, Windows and
LDAP, to protect the application. We also support HTTPS operation for an added level of
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assurance.
NEC CAS is also used for other applications as well, including the Command Line
Interface and ACD Web Mat. This eliminates the need for the user to login again and
again.
NEC CAS can disable the user accounts if X number of invalid passwords are entered
consecutively. A notification of the account lockout can also be sent to the system
administrator via either email or screen pop.
Encryption is also used between the PBX and the Management System as well. This is
based on SSL encryption so that clear text information is not passed between the two
systems over the network.
User roles can also be put in to place to limit users to only access certain parts of the
management system. These are highly granular and allow configuration of Create,
Delete or View/Update access to certain parts of the system. They can also limit based
on PBX resource. For example, you can limit a user to only see stations 1000-1020 and
then limit them to only make changes to the call forwarding and the last 2 buttons on the
phone itself.
Audit and Alarm history are an integral part of the overall security solution as well.
MA4000 can track all changes made to the system in an audit log that can tell you who
did what to the system, what did it affect, what command was run with what parameters,
and when did it occur. Also, all alarms generated by the network that MA4000 manages
are also stored on the system and can be configured to generate notifications to the
administrator.
Denial of service based attacks can be thwarted with a firewall. MA4000 runs on
Windows Server 2003 and uses IIS as its web interface. It works with any firewall
software that supports the Windows environment. Firewall configurations are detailed in
the MA4000 Security Best Practices Guide that is included with the system.
The MA4000 can also support centralized authorization codes throughout the NEC
network. When these codes are dialed, the MA4000 will check the validity of the code
against its own database and pass the authorization back to the PBX. If dialed
incorrectly, the MA4000 also has the capability of disabling the phone and generating a
notification to the administrator via either email or screen pop. This can be set up to
disable the phone after X number of invalid attempts within X seconds.
NOTE to TEQConsult – RFP Paragraph 6.0.4 not found.
6.0.5 User Interface & Tools
The management system should be operated using by GUI tools, formatted
screens, pull down menus, valid entry choices, templates, batch processing &
transactions scheduling, and database import/export. In general you should
support a user interface set for each functional area: fault, Configuration,
Performance and Security. The constituent users of each of these areas are
distinct and your interface for each should optimize the experience for that
constituent group. Management applications my integrate information from
several management areas to enhance one functional area being managed.
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NEC Response: The MA4000 Management System is a web based application that
uses friendly GUI wizards to program all major aspects of PBX administration. MA4000
supports transaction scheduling, import/export, bulk provisioning, valid entry choices,
and template based programming.
MA4000 has the following distinct sections.
Home Home Page
My Settings
Logout
Administration –
System Manager logins and roles
Assistant logins and roles
MA4000 configurations
MA4000 Assistant Configurations
Org Level setup
Alarm Configurations and notifications
Auth Code Configurations
Web Portal Configurations
System Health thresholds and notifications
Web Update configurations
PBX Configurations and Sync configurations
PBX Hardware configurations
PBX Backup schedules
Voice Mail System Configurations and Sync configurations
LDAP Server Integration configurations
Command Line Interface
Users and Devices Extension, user account and mailbox configurations
Group Programming
Template Management
Range Programming Tool
Traffic Traffic Type definitions
Traffic collection schedules
Traffic thresholds
Grade of service configurations
Traffic reports
Traffic data purge configurations
Utilities Import/Export utilities
Real Time Monitoring Tool
Configuration/audit/DESI Reports
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Audit/Alarm History
Scheduling / offline provisioning tasks interface
Help Version/License information
Online Help System
Help Search utility
PBX Manuals
ACD Web MAT Tenant management
Logon management
Position management
Trunk group management
Analog management
Holiday calendar management
Communications data management
Tenant setting management
Split management
Pilot management
CCV management
IVR management
Week Schedule Management
Holiday Schedule Management
User Settings Management
Time Out Setting Management
6.1.0 Administration Functions
The proposed systems management solution must support: station user
moves, adds, and changes; trunk group definitions and individual trunk
circuit programming; voice terminal parameters; call restriction assignments;
class of service definitions and assignments; password resets; customer
profile database; ARS routing tables; group definitions and assignments; first
digit tables; dial plan; feature access codes; paging/code call zone
assignments.
Vendor Response Requirement
Confirm the proposed systems management solution supports each of the
listed administrative functions. Identify any functions not supported.
NEC Response: Confirmed. All required functions are supported.
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Group Assignments
The administration subsystem must support each of the following group
definitions and assignments
•
Abbreviated Dialing (System, Group, Enhanced)
•
Hunt Groups
•
Call Coverage Answer Groups
•
Pickup Groups
•
Intercom Groups
•
Terminating Extension Groups
•
Trunk Groups
Vendor Response Requirement
Confirm administration support for each of the listed group definitions. List
any and all groups not supported by the administration subsystem.
NEC Response: Confirmed. All required functions are supported.
6.1.2 Facilities Performance Management & Reports
The management system must be able to collect, analyze, and provide
reports for a variety of system operations.
NEC Response: The MA4000 Management System provides a full range of on-demand
report performance and management reports.
REAL-TIME MONITORING TOOL: The Real-Time Monitoring Tool collects IP Quality of
Service (QoS) statistics directly from the NEC Univerge® SV7000 and Univerge NEAX®
2400 IPX equipment. View the status of any phone, trunk or connection trunk on any
PBX from a single screen. And receive proactive, threshold-based alerts about problems
as soon as they appear on the network.
AUDIT HISTORY: The MA4000 maintains detailed records of all system transactions so
you see exactly what changed, who changed it, and the results of the changes.
6.2.1 Basic Trunk Usage and Traffic
Trunk traffic records should be kept for all inbound and outbound calls,
identifying the trunk group and trunk channel, time and duration of call.
Vendor Response Requirement
Confirm that the proposed facilities management system satisfies this
requirement.
NEC Response: All required functions are supported within the MA4000.
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6.2.1.1 Individual Trunk Line Counters
Vendor Response Requirement
Confirm that individual trunk line counters measure and report: Number of
call attempts; Number of blocked trunk lines; Traffic intensity (Erlangs).
NEC Response: All required functions are supported within the MA4000.
6.2.1.2 Outgoing Trunk Route Counters
Vendor Response Requirement
Confirm that outgoing trunk route counters measure and report: Number of
outgoing attempts; Number of successful calls overflowing to another route;
Number of lost calls due to blocking; Number of blocked trunks in
measurement; Traffic intensity (Erlangs).
NEC Response: All required functions are supported within the MA4000.
6.2.1.3 Incoming Trunk Route Counters
Vendor Response Requirement
Confirm that incoming trunk route counters measure and report: Number of
incoming call attempts; Number of trunks in the measurement; Number of
blocked trunks in the measurement; Traffic intensity (Erlangs).
NEC Response: All required functions are supported within the MA4000.
6.2.1.4 Both Way Trunk Route Counters
Vendor Response Requirement
Confirm that both way trunk route counters measure and report: Number of
incoming call attempts; Number of trunks in the measurement; Number of
blocked trunks in the measurement; Traffic intensity (Erlangs).
NEC Response: All required functions are supported within the MA4000.
6.2.2 Attendant Consoles
Attendant counters should measure all attendants in the system, or
individual attendant positions. Record measurements include: number of
answered calls; number of calls initiated by attendant; accumulated handling
time for all calls; accumulated handling time for recalls; accumulated
handling time for calls initiated by attendant; accumulated total delay time
for recalls; number of answered recalls; number of abandoned attendant
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recalls; accumulated waiting time for abandoned calls to an attendant;
accumulated waiting time for abandoned recalls, and accumulated response
time for all types of calls.
Vendor Response Requirement
Confirm that attendant counters measure and provide reports for each of the
listed parameters. Identify attendant parameters which are not measured.
NEC Response: At present, this would be supported by use of the MTS Call Accounting
package integrated with the MA4000 via Station Call Records. This would only collect
completed calls and not call attempts, blocked calls or abandoned calls.
Additionally, NEC has included the OpenWorX Attendant Statistics software in this
proposal. OpenWorX Attendant Statistics runs on the OpenWorX platform in conjunction
with the Business Attendant System. Attendant Statistics enhances the value of the
Business Attendant System by providing statistical reports on queue activity and
attendant call processing activities.
This application gathers call data and attendant transactions and stores this information
in a database on the OpenWorX server. This data is used to generate scheduled reports
or reports on demand. A web interface, accessible from any computer with network
connectivity to the OpenWorX server, is provided for generating, scheduling, viewing
and printing reports.
Statistics on the number of calls queued, abandoned and answered along with wait
duration and call duration can be invaluable information for business planning, allowing
the organization to project the resources required to handle calls to and from attendant
positions.
In addition, statistics on the activities of each attendant – calls made, calls handled, talk
time, idle time, login and logout times – provide information for attendant performance
evaluation.
6.2.3 Stations
Station counters should measure individual stations or station group traffic
statistics, including: number calls; number of stations in measurement;
number of blocked stations in measurement; traffic rating (Erlangs).
Vendor Response Requirement
Confirm that station counters measure and provide reports for each of the
listed parameters. Identify station parameters which are not measured.
NEC Response: All required functions are supported within the MA4000.
6.2.4 Traffic distribution
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When applicable, traffic distribution across the internal switching network
should be measured for each local TDM bus, traffic over each highway bus,
and traffic across the center stage switch by each switch network interface
link.
Vendor Response Requirement
Confirm that traffic distribution is measured and reported for each switch
network element listed. Identify what is not measured and reported.
NEC Response: All required functions are supported within the MA4000.
6.2.5 Busy hour traffic analysis
Busy hour traffic analysis measurements for trunks, stations, and the internal
switch network should be performed and reported for any one hour interval
for any time of the day.
Vendor Response Requirement
Confirm busy hour traffic measurements for trunks, stations, and the internal
switch network for any one hour interval for any time of the day.
NEC Response: All required functions are supported within the MA4000.
6.2.6 Erlang Ratings
Erlang rating should be calculated and reported for individual trunk lines,
each trunk group, and all trunk groups. CCS ratings should be calculated for
individual stations or groups of stations.
Vendor Response Requirement
Confirm Erlang and CCS rating calculations and reporting for each listed item.
NEC Response: All required functions are supported within the MA4000.
6.2.7 Processor Occupancy
System call processing performance is measured in terms of Busy Hour Calls
(Attempts and Completions). The percent of maximum call processing
capacity should be reported for programmed time intervals. Threshold
reports should also be generated to monitor system load factors.
Vendor Response Requirement
Confirm measurement and reporting of processor occupancy and threshold
levels
NEC Response: All required functions are supported within the MA4000.
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6.2.8 Threshold Alarms
For a variety of system hardware devices it should be possible to define a
congestion threshold value, and measure generated alarms. Alarms are
recorded in an Alarm Record Log. The types of devices that can be tracked
include: tone receivers; DTMF senders and receivers; conference bridges;
trunk routes; modem groups.
Vendor Response Requirement
Confirm recording and reporting of alarms for each listed item.
NEC Response: All required functions are supported within the MA4000.
6.2.9 Feature Usage
Feature usage counters for selected station features, e.g., call forward, call
transfer, add-on conference, and attendant system features, e.g., recall,
break-in, should be measured and reported for programmed intervals.
Vendor Response Requirement
Confirm recording and reporting of feature usage counters for both station
and attendant operations.
NEC Response: Not currently supported via a GUI interface but could be supported via
a craft interface or using the Command Line Interface of the MA4000.
Future enhancements will allow the MA4000 to collect usage, traffic and peg counts
directly from the voice servers.
6.2.10
VoIP Monitoring
The management system should collect and store data to track usage and
performance data of IP gateway devices, IP phones, and VoIP intercom/trunk
calls. VoIP information reports may include: tracking of IP gateway devices
and calls that pass through each gateway; gateway congestion; assignment
of services or routes to gateways; tracking of phone numbers dialed or
originating off-site numbers; and IP gateway addresses.
Vendor Response Requirement
Briefly describe all VoIP monitoring information records and reports that are
available. Specify if VoIP QoS parameters such as jitter, call delay/latency,
and packet loss are tracked and reported, and if a system administrator can
monitor VoIP calls in real-time for QoS observing? Indicate if any third party
equipment is being proposed as part of your solution.
NEC Response: MA4000 has the capability of monitoring the quality statistics and IP
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traffic flows of the VoIP devices within the NEC network. This includes stations, SIP
devices, IP Trunks, IP PAD devices, Media Gateways, Conference Gateways, etc…
6.3
Optional Reports
Directory records may include each subscriber’s name along with a variety of
phone numbers such as primary, published, listed, emergency, and alternate,
as well as authorization code information, job title, employee number,
current employment status and SSN.
Inventory records and management is used to administer any kind of
inventory product part, including: PBX common equipment (cabinets,
carriers, circuit cards); voice terminals and module options; jacks, and
button maps. The reports allow administrators to accurately re-charge items.
Inventory can be tracked by data such as user, system (PBX or other
networks), jack, serial number, asset tags, trouble calls, recurring and nonrecurring costs, and general ledger codes. The inventory management
system may also include records containing the following data: purchase
date, purchase order number, depreciation, lease dates, manufacturer and
warranty information.
Cabling records keep track of all cable, wire pairs, distribution frames, wiring
closets and all connections (including circuits) down to both the position and
the pair level. Cable records include starting and ending locations,
description, type and function. Individual cable lengths are maintained and
automatically added, as is the decibel loss, for the entire path. Information
can also be provided on the status of all cable runs, as well as the number of
pairs it contains, the status of the pairs, and the type of service it provides.
Vendor Response Requirement
Identify and briefly describe your proposed management system’s Directory,
Inventory, and Cabling reports, if available.
NEC Response: These items would be supported via the MTS system integrated to the
MA4000 with addition of its Asset and Cable Management options.
6.4.0 Call Detail Recording
Call Detail Record (CDR) data should be compiled for all successful incoming
and outgoing trunk calls. Call record fields typically include the following:
•
•
•
Date
Time
Call Duration
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•
•
•
•
•
•
•
•
•
•
•
•
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Condition Code (categorizes information represented in the call record)
Trunk Access Codes
Dialed Number
Calling Number
Account Code
Authorization Code
Facility Restriction Level for Private Network Calls
Transit Network Selection Code (ISDN access code to route calls to a
specific inter-exchange carrier)
ISDN Bearer Capability Class
Call Bandwidth
Operator System Access (ISDN access code to route calls to a specific
network operator)
Time in Queue
Incoming Trunk ID
Incoming Ring Interval Duration
Outgoing Trunk ID
Vendor Response Requirement
VoiceCon will purchase its own third party call accounting and billing system.
Identify all available CDR reports that can be generated for any or the entire
call record field data listed above.
NEC Response: The following fields are available as call record output from the Voice
Server:
Date
Time
Call Duration
Condition Code (categorizes information represented in the call record)
Trunk Access Codes
Dialed Number
Calling Number
Account Code
Authorization Code
Incoming Trunk ID
Outgoing Trunk ID
6.5.0 Maintenance
System maintenance operations should, at minimum, support the following:
Monitoring of processor status; Monitoring and testing of all port and service
circuit packs; Monitoring and control of power units, fans, and environmental
sensors; Monitoring of peripherals (voice terminals and trunk circuits);
Initiate emergency transfer and control to backup systems; Originate alarm
information and activate alarms.
Vendor Response Requirement
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Confirm support of each listed maintenance monitoring activity. Identify any
activity not supported.
NEC Response: All required functions are supported within the MA4000.
6.5.1 Alarm Conditions
There are usually several types of communications system alarm conditions:
Major, Minor, and Warning.
Vendor Response Requirement
Briefly describe how your management system defines a Major, Minor, and
Warning alarm.
NEC Response: A Major Alarm is a condition which degrades system operation to the
extent that calls can not be completed or are cut-off. Minor Alarms do not result in
disconnections but represent some item which causes system degradation.
Informational alarms are just what they sound like.
6.5.2 Maintenance Reports
Vendor Response Requirement
Identify any and all available maintenance alarm reports provided by your
management system.
NEC Response: All Fault Messages from PBX, Hard Disk usage on MA4000 Server,
Reports or scheduled tasks that fail, LDAP schema invalid, LDAP synch failure and
more. Alarms may also be collected from the different subsystems of the MA4000 itself
such as for database or hard drive usage or service failure. There are also alarms
generated for security violations such as invalid auth code requests or invalid
management system logins. Traffic thresholds and VoIP quality thresholds may also
generate alarms. Reports may be filtered by type of alarm, severity of alarm, date or
time.
6.5.3 Remote Maintenance
Vendor Response Requirement
Briefly describe the available options used to support remote maintenance
operations for both customer access and for an outside maintenance service
provider. Specify how the system alerts a remote service center when an
alarm condition occurs, the trunk circuit requirements for alert transmissions,
and security measures to prevent unauthorized access.
NEC Response: Access for customer and technician is via a Web Browser interface.
Alarm Notification can be sent via MA4000 by PC Screen Pop or via email. Alarms can
be sent direct from voice server by SNMP.
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6.6.0 Provisioning
All services should be provisioned in one step. Services should include
station configuration, voice mailbox configuration, E-911 location, billing
attributes, directory attributes, and mobile Email attributes (Blackberry) and
the configuration of other end user applications.
For example, if your solution includes a zone paging application, the ability to
assign a station to a zone and change the zone membership as a whole must
be accessible through the configuration (provisioning) interface.
Templates must be supported to organize different settings across different
systems according to organizational need. At a minimum, the voice station
configuration and the associated voice mailbox must be provisioned in one
step through one interface.
Your proposed provisioning application or interface must create a complete
audit trail and must allow groups of changes to be scheduled for a future
time. Further, the solution must support mass create, delete and modify
functions to support bulk operations.
Vendor Response Requirement
Describe the provisioning workflow you recommend showing how each of
your proposed solution components is utilized. List any functions above
which are not available. List any systems or devices which are not now part
of your provisioning interface and provide a roadmap statement of how you
will treat this situation going forward.
NEC Response: All required functions are supported within the MA4000.
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Provisioning Flow within the MA4000
The MA4000 management system integrates with all of the other applications that
integrate to the NEC PBX. This integration will provide all configuration management
and provisioning change information out to the various applications once they are
changed in the MA4000.
An example provisioning flow is as follows:
The new user is created in the customer service group. The administrator would
go to the MA4000 to create this new user. They would use the “Range Add” tool
to first select a template and PBX. The MA4000 would then ask them to either
pick one of the available station numbers or it will allocate one automatically. If
the station to be created is an IP station, then MA4000 will select a port
automatically. If it is a TDM station, then MA4000 will ask the administrator to
select from a list of available ports or it will choose one automatically. The
administrator is then asked to enter a name for the user and MAC address of the
phone (optional). They can then go ahead and create that user then or they can
schedule the task to happen at a later time using the MA4000 scheduling engine.
The “Range Add” tool can also be used to add a list of new phones/users as well.
Once the task is run, MA4000 will create the new phone in the PBX, create the
new mailbox in the voice mail system, and notify the Call Accounting and
Directory applications that a new user has been entered.
There are many other useful tools within the MA4000 and a few other
ways to do different types of provisioning on the PBX or voice mail
systems:
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The Range Programming Tool:
The range programming tool is a bulk provisioning engine that is tied in with the MA4000
scheduling engine. All of these tools work for both the PBX and the Voice Mail system.
From this tool the administrator is offered the following options:
Range Programming Administrator Options
Bulk add tool based on templates
Add
Delete 1 or more stations (automatically removes from groups and
Delete
appearances)
Change any phone or mailbox configuration across multiple stations.
Change
Also change make busy status.
Change station or mailbox number
Renumber
TDM function. Move from one port to another
Move
Copy provisioning from one phone/mailbox to another
Copy
TDM function. Swap 2 or more stations
Swap
Import/Export Utility:
The import utility allows the administrator to import a list of names and station numbers
from a .CSV file and then apply a template to the list to either update or create new
stations.
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LDAP Auto Provisioning:
Audit History:
All details about every transaction that has taken place on the system is
tracked within the MA4000 Audit History database. The audit system
tracks who did what, what did it effect, when did it happen, and any
commands and parameters associated with those commands that were
issued to the PBX.
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7.0.0 Integrated Messaging System
VoiceCon requires a HQ-based voice messaging system that must be fully
integrated with the proposed IPTS network solution. VoiceCon also requires
integration of the proposed voice messaging system with a Microsoft
Exchange messaging system to provide “unified” messaging applications. The
proposed voice messaging system solution must be centrally located at the
VoiceCon HQ location, and be capable of supporting station users at all
remote VoiceCon faciliities (RO and SBs).
The voice mail system will also serve as an automated attendant position for
select incoming trunk calls, and also as a secondary point of coverage as an
automated attendant system for designated stations. All software and
hardware necessary to interface with the existing telephone system will be
provided under this bid.
The sizing requirements are:
Installed/Equipped Capacity Maximum Capacity
Number of Users
Number of Ports
Hours of Storage
2000
64
1000
3,000
96
1200
Five (5) automated attendant ports are included in the requirements. A
Grade of Service level of P.01 is required.
Vendor Response Requirement
Briefly describe the proposed integrated messaging solution, and provide
details about the voice mail system architecture and it’s interconnection to
the voice communications system and Microsoft Exchange system. Include
processing system platform information in the discussion. Verify that the
system being bid can comply with each of the proceeding requirements.
NEC Response: NEAXMail AD-120 can be configured in several ways—from a standalone voice-messaging server to a unified messaging server connected to an Exchange
network. NEAXMail AD-120 servers can be clustered to provide a high-capacity
redundant system that can continue to operate even if a particular server fails. The
NEAXMail AD-120 system achieves an excellent balance between tight integration with
Exchange and the rapid performance of a SQL database. All configurations of NEAXMail
AD-120, including the stand-alone voice-messaging server, use Exchange for address
directory information and for message storage. NEAXMail AD-120 uses LDAP
(Lightweight Directory Access Protocol) to access address information from Active
Directory and the Exchange directory. This information is then cached in an SQL server
database to provide rapid access to directory information. Additionally, settings for
subscribers, call handlers, interview handlers, location objects, and other NEAXMail AD-
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120 entities are all stored in the SQL server database. NEAXMail AD-120 uses MAPI
(Mail Application Programming Interface) to access voice and fax messages in the
Exchange message store. By using Exchange, NEAXMail AD-120 has access to the
same address directory and message store used by e-mail clients. By using SQL Server,
NEAXMail AD-120 provides excellent performance. By using LDAP, NEAXMail AD-120
keeps Active Directory, Exchange and SQL data synchronized, without requiring any
additional administrative effort. NEAXMail AD-120 integrates with Exchange 2000 and
2003.
7.1.0 Support for Open Standards
Vendor Response Requirement
Describe voice messaging system’s support for open standards.
List the clients that can be used with your proposed solution.
For proprietary clients, detail minimum hardware and software
requirements
NEC Response: The NEAXMail AD-120 unified messaging server is built on a
foundation of enterprise operating systems and applications developed by Microsoft.
Taking advantage of open standards such as MAPI, IMAP, ADSI, and LDAP provide
core functionality today, while allowing us to prepare for the features of tomorrow. The
NEAXMail AD-120 integrates with the Microsoft desktop clients—including the Exchange
Inbox, Outlook 98, 2000, 2002, and 2003.
7.1.1 Security Features
Vendor Response Requirement
Describe security features available with the voice messaging system to
prevent abuse and unauthorized access.
NEC Response: The NEAXMail AD-120 has advanced security features that ensure the
safety and integrity of the unified communications environment by offering many
advanced security features, such as the ability to detect hackers, lock account, set
password policy, and force password reset on the next login.
Hacker Detect and Lock Accounts - NEAXMail AD-120 monitors the number
of attempts made to log on to an account via the telephone and can lock
accounts if incorrect passwords are entered repeatedly. The system
administrator can specify the number of invalid log-on attempts allowed and
the number of minutes before the system resets the account lockout. Locked
accounts can be unlocked manually by the system administrator or
automatically by NEAXMail AD-120 after a specified number of minutes.
Passwords - the system administrator can define the number of characters
for passwords, historical password tracking and how often NEAXMail AD-120
requires a password to be changed. By determining the rules for subscriber
account policy, the system administrator customizes NEAXMail AD-120 for
an organization’s specific needs.
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7.2.0 Voice Mail Features
7.2.1 Forwarding
The system must provide access for forwarded calls from:
* Customer telephone system
* Direct central office (Business or Centrex lines)
* 800 Service lines
Vendor Response Requirement
Confirm support for each forwarding requirement.
NEC Response: The NEAXMail AD-120 supports all the forwarding requirements
above.
7.2.2. Disconnect Detection
The system should detect that a caller has hung up and immediately
disconnect and restore the line to service.
Vendor Response Requirement
Confirm support for this operation.
NEC Response: The NEAXMail AD-120 supports disconnect detection requirements
above.
7.2.3. Station Dialing
In addition to the menu/route, callers may access an individual station either
through the input of the extension number or the input of the called party's
last name. A total of 2,000 names plus 100 extension numbers will be
possible.
Vendor Response Requirement
Confirm support for this operation.
NEC Response: The NEAXMail AD-120 supports the station dialing requirements
above.
7.2.4 Answer Announcement
Individual, personalized announcements of 15-30 seconds for each mailbox
user will be possible. A user's dictated answer message will only occupy the
number of seconds dictated, with the remainder to be pooled so as to be
available to:
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1) all other mailbox owners; and,
2) for message taking.
A system announcement of up to 30 seconds will be possible and also will be
available in the event of switching system failure. It will be possible for the
mailbox owner to input separate greetings for calls received internally or
externally on the system. It will be possible for several individuals to share
the same mailbox extension number. A caller reaching such a mailbox will be
able to select between individual mailboxes.
Vendor Response Requirement
Confirm support for these operations.
NEC Response: The NEAXMail AD-120 supports all the answer announcement
requirements above.
7.2.5 DTMF Signaling
The system will be capable of receiving and generating standard
DTMF tone signaling.
Vendor Response Requirement
Confirm support for this feature.
NEC Response: The NEAXMail AD-120 supports the DTMP Signaling requirement
above.
7.2.6 Greeting
Voice mail calls will be answered on the first ring and be time- and datestamped.
Vendor Response Requirement
Confirm support for this feature.
NEC Response: The NEAXMail AD-120 supports the greeting requirements above.
7.2.7 Escape
A caller reaching the voice mail system will have the ability to re-route to an
extension by dialing up to five digits or the operator by dialing "0" before or
after leaving a message. It will not be possible for a caller reconnected to the
telephone system to be connected to the public network.
Vendor Response Requirement
Confirm support for this feature.
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NEC Response: The NEAXMail AD-120 supports the escape requirements above.
7.2.8 Trunk Access
It will be impossible for a caller passing through the attendant to reach an
outside line.
Vendor Response Requirement
Confirm support for this feature.
NEC Response: The NEAXMail AD-120 can support this requirement assuming proper
system administration has been done.
7.2.9 Distribution Lists
The system will contain a minimum of 80 distribution lists of at least 25
names each plus "all broadcast."
Vendor Response Requirement
Confirm support for this feature.
NEC Response: The NEAXMail AD-120 supports the distribution lists requirements
above.
7.2.10 Message Forwarding
Messages may be forwarded to single or multiple destinations with or without
introductory comments.
Vendor Response Requirement
Confirm support for this feature.
NEC Response: The NEAXMail AD-120 supports the messaging forwarding
requirements above.
7.2.11 Audit Trail
It will be possible for a user to designate a necessary written record of
message destination, input time and receipt. This audit trail will be printed on
the administrative console together with daily reports.
Vendor Response Requirement
Confirm support for this feature.
NEC Response: The NEAXMail AD-120 supports the audit trail requirements above.
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7.2.12 Message Indication
The receipt of a message in a mailbox will cause a message-waiting lamp or
"stutter" dial tone upon lifting of the station handset to indicate a messagewaiting condition.
Vendor Response Requirement
Confirm support for this feature.
NEC Response: The NEAXMail AD-120 supports the message indication requirements
above.
7.2.13 Identification Code
Users accessing the
system will input a discrete six-digit identification code which will be
positively validated prior to access to their mailbox. Identification codes
may be changed by mailbox owner.
Vendor Response Requirement
Confirm support for this feature.
NEC Response: The NEAXMail AD-120 supports the identification code requirements
above.
7.2.14 Message Recovery
The mailbox owner accessing the mailbox will be automatically told how
many new messages have been received since last access and how many
saved messages exist. Upon accessing the messages, the subscriber will
have the choice of deleting, skipping or saving a message. Saved messages
may only be deleted by the subscriber or by the system administrator.
Vendor Response Requirement
Confirm support for this feature.
NEC Response: The NEAXMail AD-120 supports the message recovery requirements
above.
7.2.15 Message Reply
A mailbox owner may respond to a message input by another system
mailbox owner by simply depressing a single key.
Vendor Response Requirement
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Confirm support for this feature.
NEC Response: The NEAXMail AD-120 supports the message reply requirements
above.
7.2.16 Message Review
It will be possible for a user to review and edit either an announcement or
input a message.
Vendor Response Requirement
Confirm support for this feature.
NEC Response: The NEAXMail AD-120 supports the message review requirements
above.
7.2.17 User Controls
A user accessing their mailbox will be capable of the following control
functions:
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
Playback messages
Skip to next message
Cancel review
Replay last message
Replay faster or slower
Pause
Append information
Forward message (to mailbox or list)
Create new answer announcement
Increase play-back volume
Vendor Response Requirement
Confirm support for this feature. Indicate if any function is not supported.
NEC Response: The NEAXMail AD-120 supports the user controls requirements above.
7.2.18 System Management Console
The system will be equipped with a CRT and printer to provide system
management functions. The administrative programs and traffic information
secured will be possible during system operation. Traffic reports will be
available on customer demand or automatically on a pre-programmed basis
in quarter, half or one hour time frames or daily and weekly. At a minimum,
they will indicate the following:
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1. Storage space used for announcements or information mailboxes.
2. Storage space used for messages.
3. Maximum storage space used during the interval.
Vendor Response Requirement
Confirm support for this feature. Indicate if any requirement is not
supported.
NEC Response: The NEAXMail AD-120 supports the system management console
requirements above, but it does not support scheduling reports in advance.
7.2.19 Traffic Reports
Traffic reports will be available on customer demand or automatically on a
pre-programmed basis in quarter, half or one hour time frames or daily and
weekly. At a minimum, they will indicate the following:
1. Storage space used for announcements
2. Total calls answered
3. Total calls routed to station
4. Total calls routed to default
5. Total calls abandoned
6. CCS use and call count by input
Vendor Response Requirement
Confirm support for this feature. Indicate if any requirement is
not supported
NEC Response: The NEAXMail AD-120 supports the traffic report requirements above,
but it does not support scheduling reports in advance.
7.2.20 System Changeability
It will be possible for the system administrator to add and/or delete
mailboxes, change general recordings and perform other administrative
duties while the system is in operation.
Vendor Response Requirement
Confirm support for this feature
NEC Response: The
requirements above.
NEAXMail
AD-120
supports
the
system
changeability
7.3.0 Networking
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VoiceCon plans on networking it new HQ messaging system to
other VoiceCon locations equipped with messaging systems.
7.3.1 AMIS
The proposed messaging system should support AMIS networking standards.
Vendor Response Requirement
Confirm support for these features
NEC Response: The NEAXMail AD-120 supports the AMIS requirements above.
7.3.2 Digital IP Networking
The proposed messaging system should support VPIM networking standards.
Vendor Response Requirement
Briefly describe digital networking capabilities of your proposed messaging
system solution. Indicate if VPIM is supported.
NEC Response: NEAXMail AD-120 supports two types of digital networking to transmit
messages between other AD-120 systems or other vendor systems. VPIM (Voice
Profile for Internet Mail) is utilized for message delivery and communication between
other manufacturers’ voice mail systems supporting this protocol. ActiveNet is a
proprietary digital networking protocol that enables the NEAXMail AD-120 to connect
with other NEAXMail AD-120 servers at remote sites via a WAN or the Internet.
7.4
Integrated Messaging Application
Vendor Response Requirement
Briefly describe how the proposed voice messaging system is to be
integrated with VoiceCon’s text messaging system, based on a MS Exchange
server, to provide unified messaging system functionality. Station users
must be able to view and access all messages (voice, text, fax) from their PC
display monitor. Email text messages must be accessable from a telephone
using text-to-speech conversion.
NEC Response: NEAXMail AD-120 delivers true unified messaging via Microsoft
Exchange and the ViewMail for Microsoft Outlook (VMO) form for better access to, and
management of, all of a subscriber’s voice, fax and e-mail messages. Integrating with
the Microsoft desktop clients—including the Outlook 98, 2000, 2002, and 2003 – the
NEAXMail AD-120 provides an intuitive graphical user interface accessible from any
networked PC, and advanced Text-to-Speech features from any touch-tone telephone.
With just a click of the mouse, subscribers access voice, fax, and e-mail messages, and
reply, forward, and save them in public or personal folders within Exchange/Outlook. The
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icons accompanying those messages make it easy to distinguish between voice, fax,
and e-mail communications, saved and new messages, and the priority (normal, urgent,
private) with which they were sent. Faxes can be viewed on screen and printed from any
networked PC, or forwarded to any fax machine from a touch-tone telephone.
Subscribers can download all types of messages and work with them off line, and apply
Inbox/Outlook Assistant rules to streamline communications management. NEAXMail
AD-120 unifies traditionally disparate communications methods so employees can work
more efficiently.
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8.0.0 Contact Center (Informational, only)
VoiceCon has future plans to install and operate a mid-size contact center
solution across its three HQ locations. The contact center would integrate
incoming voice, email, and web contacts from customers, and also support
outgoing voice calls to potential customers. It is anticipated that the contact
center will require 50 multifunction agents positions, and 5 supervisor
positions. The contact center features and functions are NOT to be
included in the configuration or pricing proposal.
8.1.0 Incoming Voice Call Center
The voice contact center solution should support call prompting, detailed call
screening, and intelligent call routing capabilities. Agent groups should be
both fixed and virtual based on skill profiles of the agents. Client/server CTI
applications must be supported at all agent desks. Agent group assignments
must be able to be distributed across the three HQ locations. The system
should be designed to minimize agent requirements and call waiting times.
Realtime supervisor reports and detailed historical reporting is required.
Vendor Response Requirement
Briefly describe you’re the system architecture of your incoming voice call
center solution to satisfy VoiceCon’s basic requirements (see below). Include
specific information about the system design architecture of your solution
(hardware and software requirements), and specific capacity parameters for
agents, supervisors, groups, announcements, queue slots, trunks and trunk
groups, et al.
NEC Response: The ACD software is an integral part of the SV7000. Detailed
information about system design architecture is provided in response to the specific
response requirements (see below).
8.1.1 Basic Call Control Capabilities
At a minimum the proposed solution must be able to provide call control
based on:
• ANI/DNIS
• call volumes
• performance criteria
• priority queuing
Vendor Response Requirement
Briefly describe the call control methodology used by your system that
analyzes, routes, and queues calls based on each of the criteria.
NEC Response: The Genesys routing engine defines all objects within the Contact
Center as data attributes. Once they are defined, they are loaded into a real-time server
process/engine which allows the Genesys Strategy Designer to call upon those objects
as it becomes necessary. Thus as we create the abstraction layer away from the overall
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hardware architecture, we allow the business to define what variables are most critical
for their business. This abstraction process not only allows for several different standard
criteria’s to be used, such as call volumes, priority, queue depth, talk time, skills, virtual
skills or virtual route points, or any other standard Contact Center variant, but also allow
for non-traditional objects to be place into the routing queue. This allows the business to
truly run the Contact Center more to the true function of what an agent does, instead of
just creating routing strategies which are designed to only understand telephony centric
definition.
8.1.2 Advanced Call Control Capabilities
As an option the proposed solution must be able to provide call control based
on:
• agent skills
• customer preference
• inbound and outbound call levels
• multi-media
Vendor Response Requirement
Briefly describe your system’s call control methodology that analyzes, routes,
and queues calls based on each of the criteria.
NEC Response: The Genesys routing engine defines all objects within the Contact
Center as data attributes. Once they are defined, they are loaded into a real-time server
process/engine which allows the Genesys Strategy Designer to call upon those objects
as it becomes necessary. Thus as we create the abstraction layer away from the overall
hardware architecture, we allow the business to define what variables are most critical
for their business. This abstraction process not only allows for several different standard
criteria’s to be used, such as call volumes, priority, queue depth, talk time, skills, virtual
skills or virtual route points, or any other standard Contact Center variant, but also allow
for non-traditional objects to be place into the routing queue. This allows the business to
truly run the Contact Center more to the true function of what an agent does, instead of
just creating routing strategies which are designed to only understand telephony centric
definition.
8.1.3 Caller Notification of Wait Time
The proposed solution must be able to notify callers of expected wait times
and “place” in queue and support information collection (such as an
automated attendant feature) using “internal” hardware and software.
Vendor Response Requirement
Describe how the application calculates wait time and any optional hardware
or software required. Include a statement addressing if the announcement
of wait time has an impact on a caller’s state in queue?
NEC Response: Within the Genesys software architecture there is a real-time server
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process called Stat Server. The Stat Server is the server process which loads all the
defined and custom defined statistics to route interaction upon. Within the model are
Caller Wait Time statistics, Queue Depths statistics and many, many others. To provide
this functionality into a step within the routing strategy, the user simply accesses the GUI
interface, Strategy Designer, and double clicks the icon, and follows the wizard to
implement the steps.
8.1.4 Transfer to Voice Messaging Application
After a configurable time, the caller should be able to transfer to a voice
messaging system to leave a callback message.
Vendor Response Requirement
If the caller chooses to continue waiting rather than hanging up after leaving
a message, describe how the call is placed back in queue.
NEC Response: Once the voice mail is identified as a target for which the router can
place calls to, a simple menu step is added for the routing strategy to allow the customer
the choice between leaving a message or continuing to hold. Depending on the
integration from the voice mail system to provide messaging to the Genesys solution to
execute a callback, Genesys could execute the callback or simply end the call
transaction and let the voice mail or IVR application execute the callback, more
information on the voice mail system and the customer environment is required to
determine the best possible way to provide this function.
8.1.5 GUI Administration Tool
Supervisors must be able to reconfigure call control and assignments in real
time, change priority of multiple calls simultaneously, view details of
orphaned calls and retain customized settings regardless of log-on location.
The solution must use a GUI administration tool and provide a graphical
editor and what-if modeling as standard.
Vendor Response Requirement
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Describe the system’s GUI administration tools.
NEC Response: The Genesys Strategy Designer is a GUI based tool which is similar to
Visio in that it allows the user to drop and drag icons from a palate and then double click
the icon to follow a wizard’s base step. The Genesys tools allow for extensive
customization and provide a large library of standard functionality within each icon.
8.1.6 Soft Client
A soft client agent telephone and supervisor console will be highly desirable
for both premises and off-premises locations.
Vendor Response Requirement
Describe the soft clients available for agent and supervisor use. The soft
client must provide on-line help, ability to reserve calls or change call
priority. For proprietary clients, detail minimum hardware and software
requirements.
NEC Response: Genesys provides two clients which require little in the way of
hardware support, and as such are designed to work with today’s standard PC
workstation configuration, requiring as little as a Pen III, Windows Operating System,
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and 64Megs of Ram to operate.
The first of these three clients is Genesys Contact Navigator, a thick or thin client version
which provides the agent a complete multi-media soft phone which operates seamlessly
with any existing telephone instrument.
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The second client is allows the supervisor the ability to support ad-hoc routing of emails
only, Genesys Supervisor Desktop.
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Finally, the supervisor also has a GUI client which allows the supervisor in real-time to
see the Contact Center. These views are completely customizable from a wizards
based tool within the Genesys CCPulse+ window, and allows the ad-hoc views within
reach of few clicks of the a mouse.
8.1.7 ACD Voice Terminal
IP desktop voice terminal instruments will be required for agent positions.
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Vendor Response Requirement
Briefly describe any telephone instruments designed specifically for ACD
agents. Include any and all feature/function attributes unique to ACD
operations. Provide a photograph of the instrument, if available.
NEC Response: NEC proposes to use the Dterm IP 16-line display telephone to meet
this requirement. A picture of this instrument is included in the Dterm PowerPoint,
included with this response.
8.1.8 Supervisor Real-time Call Handling and Performance Status
Supervisor terminals must show, in real time, all logged-on
agents, the status of each agent, caller queue information and
thresholds and alarms. Users must be able to customize
displays.
Vendor Response Requirement
Describe the proposed solution's real time supervisor console display
capabilities for assisting supervisors with managing the customer interaction
center. Include a diagram illustrating two or three screen displays available
to the supervisor.
NEC Response: The Genesys CCPulse+ tool allows a supervisor to dice and slice any
view which he or she desires. View templates are stored on the local client and when
the supervisor logs on, they simply need to launch the view they wish to see. These
views can be archived to allow views/templates to be viewed at a later date. New views
can be create at anytime and do not require technical assistance to produce.
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8.1.9 Agent Display Information
Vendor Response Requirement Describe real-time display
information provided to agents at their desktop via their hard
telephone instrument and the softclient solution.
NEC Response: The images of CCPulse provided with this RFP response are clients
which any agent may have at their desktop, or if the customer desires, a customize
desktop toolbar could be developed which would minimize the desktop real-estate and
allow for a ticker tape presentation.
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8.2.0 Reporting
VoiceCon requires call center system operation reports in various formats.
8.2.1 Statistical and Configuration Reporting
VoiceCon requires sophisticated reporting to track and further enhance its
CIC operations. Reports must be available on terminal display and paper
printout and be able to be downloaded to a PC. The proposed solution must
provide open storage capability.
Vendor Response Requirement
Describe the number of and type of information standard statistical,
configuration and audit reports provided.
NEC Response: The Genesys Call Center Analyzer provides sixty four standard out-ofthe-box reports which are done through the Brio presentation layer. The reporting
architecture is modeled after a standard data warehouse model and thus any amount of
data stored is only limited to the size and configuration of the server for which the
software is native.
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8.2.1 Graphical Reporting
The proposed solution must provide graphical reports as a standard feature.
Vendor Response Requirement
Describe the available graphical reports with your system.
NEC Response: See question 8.2.1(Statistical and Configuration Reporting).
8.2.2 Call-by-Call Reporting
The proposed solution must provide call-by-call reporting as an optional
feature.
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Vendor Response Requirement
Describe your system’s call-by-call reporting capabilities, if available.
NEC Response: Genesys provides this as a standard function of Call Center Analyzer.
8.3.0 Self Service
The proposed solution must support self service (e.g., IVR) integration as an
option. Callers must be able to retain their place in queue while using IVR
features
Vendor Response Requirement
Describe your system’s ability to support inbound calling, call control
services, messaging for agents, speech recognition, text-to-speech, TDD and
CTI and integration with a customer self-service interaction application.
NEC Response: Genesys Voice Portal was one of the first true vXML open platform,
open vXML 2.0 standards compliant voice platform. Genesys sits on the vXML
consortium which helps define the evolving vXML standard and as such wrote the
upcoming ccVXL standard which will allow speech enabled application to provide 1st
and 3rd party call control. Because the Genesys GVP is built upon the Genesys
Customer Interaction Management, (CIM) platform, CTI, routing, reporting, Workforce
Management and Self Service applications are all being driven by the same routing
engine. This simplifies the reporting, critical to a true integrated Contact Center.
8.3.1 Script Development
Vendor Response Requirement
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Describe the design tools/environment for IVR script development, the
method used to test applications and changes prior to putting them into
production and the method of putting changes into production.
NEC Response: Genesys provides the Genesys Studio tool as a GUI interface for script
development. This tool vXML compliant interface, and thus if the user has an alternative
vXML scripting tool they wish to utilize, that tool will work with Genesys GVP.
8.4.0 Workforce Management System
The proposed solution must provide forecasting and scheduling capabilities
as an option.
Vendor Response Requirement
Describe your system’s workforce management capabilities.
NEC Response: Genesys provides a complete Workforce Management tool which
provides forecasting, scheduling and adherence. The Genesys WFM tool has all the
standard feature sets that traditional WFM tools has, with one very critical difference.
When using the Genesys WFM tool within a Genesys environment, you automatically
utilize the same GUI management tool, diagnostic tool, and reporting tool. Most
importantly, when using the Genesys WFM tool, you have a direct integration to the
Genesys routing engine. Thus before a call is routed to any agent within a potential
target, the Genesys routing engine will look at the schedule and adherence reports in the
WFM. Based on customer define business rules the router may or may not route a call
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to an agent. If you are abiding to adherence scheduling, then having a seamless
integration between the routing engine and workforce management is a must.
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8.5.0 Integrated Email Call Control
The proposed solution must integrate customer email messages as an option.
It is also desirable that agents be able to handle a mix of voice and email
messages on a call-by-call basis, and that all incoming voice calls and emails
be routed into the same agent queue(s).
Vendor Response Requirement
Decribe your system’s capability to integrate email contact center functions
with your voice call center system. Include information about the hardware
and software requirements for this application.
NEC Response: The Genesys Internet Contact Solution provides a plug-in module
which allows the customer to interface to email servers via POP3 or IMAP interface. We
require no other servers other than those supporting the overall Genesys environment,
which is undefined as call volumes, email, chat and co-browse volumes are not defined.
Once this module is plugged into the Customer Interaction Management layer, the
function of routing and reporting is defined by the data objects which are contained with
in the routing engine.
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8.6.0 Web Center
VoiceCon anticipates that it will require integration of its call center with its
web server system. The proposed solution must support customer-initiated
contact through the Internet as an option.
Vendor Response Requirement
Describe how your call center can be integrated with the VoiceCon website to
allow agents to respond to customer callback requests via the website.
Include in the discussion whether agents can collaborate in realtime with
callers during an online website transaction.
NEC Response: The Genesys Internet Contact Solution provides a plug-in module
which allows the customer to interface to email servers via POP3 or IMAP interface. We
require no other servers other than those supporting the overall Genesys environment,
which is undefined as call volumes, email, chat and co-browse volumes are not defined.
Once this module is plugged into the Customer Interaction Management layer, the
function of routing and reporting is defined by the data objects which are contained with
in the routing engine. Web interaction or contact interaction are done through a defined
set of Java beans which allow a web programmer to install them into any standard web
page. This way the Genesys environment does over burden the web server, and
provides simple event exchanges from the Genesys Internet Contact Solution and the
customer’s web site.
8.7.0 Outbound Dialing
The proposed solution must support automated outbound predictive dialing
as an option.
Vendor Response Requirement
Describe your system’s capabilties to perform outbound predictive dialing,
and include necessary hardware/software requirements.
NEC Response: Genesys provides a true software only outbound dialing solution which
provides true predictive, progressive, and preview dialing. The hardware requirements
again are unknown at this time due the fact that there is no anticipated call volumes to
size servers. The real value in having a Genesys outbound dialing solution is two fold.
One, because the outbound module sits on top of the Customer Interaction Management
layer, routing and reporting are completely and seamlessly integrated into the outbound
module. Thus the ability to do true blending is standard. Additionally because it is a
software based solution, and uses the PBX switching fabric to make calls from, less
hardware is required to link the dialing application to the live agent. There is no need to
provide 2 to 1, or 3 to 1 tie-line links for predictive mode. We optimize by supervising
agents through the routing engine incoming and out going call volumes, thus bridging
calls then tearing down the unused leg of the call to make the next call.
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8.8.0 Server-Based CTI Call Control
The proposed solution must support server-based CTI applications as an
option.
Vendor Response Requirement
Describe the capabilities of the proposed solution to simultaneously route a
call and data screen populated with the caller's identity, location or reason
for calling.
NEC Response: The Genesys Customer Interaction Management layer is the
framework to support CTI screen pop. Based on the open architecture, we can and do
monitor all legs of any contact/interaction into the center, attach any relevant data to for
screen population, and store that information in our Genesys database for reporting.
This framework has a library of event messages which are gathered from any point of
contact which is used within the infrastructure, and based on an event response allows
the Genesys CIM platform to react to the that device in it native language or message
set.
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Part 2: System Pricing
1.0
System Pricing Requirements
Summary system and voice terminal pricing data will be presented to
VoiceCon workshop attendees and be deemed for public use.
Detailed pricing data will remain confidential, and used to verify if the
proposed system configurations satisfy RFP requirements.
Installation fee pricing data is required, and must be included in the RFP
response. Indicate if the proposed installation fee is based on direct
sales/service or a channel partner pricing schedule.
The proposed system price must also include a 1-year warranty to
the customer. If this is a pricing option in your pricing schedule
include it as part of the installation fee, and identify it as such.
NEC Response: The Installation Fee is based on a direct sales/service
pricing schedule.
Proposed Pricing for Installation includes all charges for the 1-year
warranty including software upgrades (bug fix and new release).
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2.0 Summary Pricing – VOICECON NETWORK (all five locations)
Complete and submit the attached EXCEL data table for your proposed
system pricing summary data. The submitted data will be made available to
the general public.
System Summary Pricing
LIST
DISCOUNTED
$306,861.44
$112,882.27
Generic Software (Standard Features)
$160,574.00
$55,459.69
Optional Software Features/Packages
$143,882.00
$49,694.54
IP Port License Fees (if applicable)
$174,136.00
$60,143.79
Desktop Voice Terminals
$870,646.00
$324,632.39
$39,009.02
$13,473.09
Messaging System
$130,964.00
$86,088.97
Installation Fee (including 1-year warranty)
$260,739.18
$243,966.38
$2,086,811.64
$946,341.13
All Common Equipment
(call processing, port interfaces, media
gateways, housings, power, feature/application
servers, et al)
Systems Management/Administration System
TOTAL
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3.0 Desktop Voice Terminal Pricing
Complete and submit the attached EXCEL data table for your proposed
Desktop Voice Terminal pricing summary data. The submitted data will be
made available to the general public.
Voice Terminals
LIST
DISCOUNTED
Economy Desktop IP Telephone Instrument
$274.00
$100.79
Administrative Desktop IP Telephone
Instrument
$450.00
$165.53
Professional Desktop IP Telephone Instrument
$690.00
$253.82
Executive Desktop IP Telephone Instrument
$880.00
$323.71
IP Audioconferencing Unit
$540.00
$307.81
PC Client Softphone (Station User) License Fee
$122.00
$42.14
$4,762.00
$1,644.72
$465.00
$171.05
PC Client Softphone (Attendant) License Fee
Key Module Add-on
Gigabit Ethernet Module Add-on (if available)
N/A
N/A
Display Module Add-on (if available)
N/A
N/A
WLAN Module Add-on (if available)
N/A
N/A
Desktop Power Module Option (if available)
$24.00
$8.83
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4.0 Detailed Configuration Components and Pricing
Submit a separate EXCEL file with a detailed listing of proposed
communications system components/elements and associated
unit pricing, also indicating the proposed unit quantities
included in the configuration for the base system (HQ facility).
Also include an additional section with the configuration
hardware/software elements and associated pricing data to
satisfy each of the remote facilities (small, medium, large).
Provide English language descriptions of all price configuration
system components and elements in addition to any proprietary
order codes.
At minimum, the configuration component list should contain:
•
•
•
•
•
•
•
•
•
•
•
•
All common control elements
All common equipment port cabinets/carriers
All port circuit interface cards for station and trunk
ports
All media gateway equipment for station and trunk
ports
All call control signaling interface cards
All voice terminals, including audioconferencing
units
Generic software
All port license fees
All optional software packages
Include all optional adjunct server equipment to
support of required features
All voice messaging system elements (cabinet
equipment and memory storage)
All systems management elements
The detailed pricing file will NOT be made public, but will be
used by VoiceCon, only, to verify adherence to system
configuration performance requirements and pricing summary
data.
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Appendices
Appendices
Detailed Features Listing
A-1 Attendant Camp-On with Tone Indication
This feature permits the Attendant, when a desired Station is busy, to hold an Incoming
Call in a Special Waiting mode. The Attendant sends a Distinctive Camp-On Tone
Indication to the Busy Station. When that Station becomes idle, it is automatically rung
and connected to the Waiting Trunk Party upon answering.
A-2 Attendant-Controlled Conference
This feature permits the Attendant to establish a Conference among as many as eight
parties. The Conferees may consist of any combination of Stations and/or Trunks,
whether Inside or Outside Parties.
A-3 Attendant Console
The ATTENDANT CONSOLE operates on a Switched Loop basis. Six Attendant Loops
terminate at each Console via the associated position circuit. The Attendant can answer,
originate, hold, extend and re-enter calls through each Loop. The number of calls may
be effectively increased to 12 through use of the ATTENDANT LOOP RELEASE
A-4 Attendant Keypad
This feature permits the Attendant to dial all calls from the ATTENDANT CONSOLE [A3] via a Pushbutton PAD.
A-5 Attendant Lockout
This feature denies an Attendant the ability to re-enter an established Trunk/Station
Connection without being recalled by the Station.
A-6 Attendant Loop Release
This feature allows an ATTENDANT CONSOLE Loop to become available for a second
call as soon as the Attendant has directed the first call to a Station, even if that Station
does not answer.
A-7 Attendant Override
This feature permits an Attendant to enter a Busy Trunk Connection within the System,
via the Attendant Console
A-8 Automatic Recall
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This feature works as a timed reminder. When a call handled by the Attendant remains
on hold, camped-on, or ringing unanswered for a fixed interval, the Attendant is
automatically alerted.
A-8D Automatic Recall - Dterm
This feature works as a timed reminder. When a Dterm handled call remains on hold
(EXCLUSIVE HOLD [E-4D] or NON-EXCLUSIVE HOLD [N-7D]) or during an unattended
transfer for a variable, programmable period of time, the Dterm is automatically alerted.
A-15 Announcement Service
This feature allows a user to hear a prearranged announcement when the user dials a
predetermined Access Code.
A-16 Alternate Routing
This feature automatically routes Outgoing On-net or Off-net Calls over alternate
facilities when the first-choice Trunk Group is busy. The user selects the first-choice
Route by dialing the corresponding Access Code, and the equipment then routes the call
through Alternate Trunk Groups only if the first is busy. The System will also add or
delete digits, when necessary, to complete the call to the desired Station.
A-17 Audible Indication Control
This feature allows the Attendant to adjust the audible indications provided to the
ATTENDANT CONSOLE
A-18 Account Code
This feature is adjunct to STATION MESSAGE DETAIL RECORDING (SMDR) [S-10],
which provides a Station User with the capability to enter a Cost-accounting or Clientbilling Code (up to ten digits) into the system before dialing a Long Distance Number.
A-19 Attendant Night Transfer
When the ATTENDANT CONSOLE [A-3] is in Night Mode, any operator calls (dial 0
calls) are automatically routed to a predetermined Night Station.
A-20 Authorization Code
An AUTHORIZATION CODE is a numerical code dialed by users (up to 10 digits), which
will override the Station CLASS OF SERVICE - INDIVIDUAL [C-15] for facility access
restriction.
The AUTHORIZATION CODE can be masked on the Dterm display if necessary using
the AUTHORIZATION CODE DISPLAY ELIMINATION [A-99].
When a wrong code is received from a Station/Trunk, detailed information on the
unauthorized user is output as System Message [26-M]. This message is Toll Fraud
Report.
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A-21 Automatic Trunk Test
The AUTOMATIC TRUNK TEST capability provides a functional test on a large number
of Trunks at a prearranged time. The results of the test are reported at the
MAINTENANCE ADMINISTRATION TERMINAL (MAT) [M-18]. The test can include
Ringback Tone Test, One KHz Test Tone and Trunk Selection Test, by having the
correct Test Trunk Termination at the Distant Office.
A-29 Authorization Code - Tie Line Incoming Routes
This feature offers an Outside Party the ability to enter an AUTHORIZATION CODE [A20] through an incoming TIE Line Trunk.
A-30 Automatic/Manual Intercom
This feature permits multiple Dterm users to simultaneously call all the Stations in a preassigned group, regardless of their idle/busy status. An INTERCOM member can
optionally override another INTERCOM Station in a Twoparty INTERCOM Connection.
This service is referred to as INTERCOM BRIDGE. INTERCOM Service is separated
into two types, AUTOMATIC INTERCOM (AICM) and MANUAL INTERCOM (MICM).
Automatic Intercom
All Stations in the INTERCOM Group are alerted simultaneously with a lamp
indication. Ringing is directed to one predefined Station, as in a button and
buzzer INTERCOM arrangement.
Manual Intercom
All Stations in the INTERCOM group are called simultaneously with a lamp
indication. The Calling INTERCOM Station manually sends a one-second tone to
the Called INTERCOM Signal Station, as in a button and buzzer INTERCOM
arrangement.
A-52 Account Code - Attendant
This feature provides the ATTENDANT CONSOLE [A-3] with the capability to enter an
Account Code into the system after talking with a Central Office Outgoing/Incoming Call.
A-53 All Zone Paging
This feature provides both the Attendant and a user with dial access to Multiple Zone
Paging Equipment.
A-59 Announcement Service - Attendant
This feature allows the Attendant, via the ATTENDANT CONSOLE [A-3], to hear a
prearranged announcement when the Attendant dials a predetermined Access Code.
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A-60 Attendant-to-Attendant Calling
This feature permits an Attendant, via an Attendant Console, to access a non-specific
free Attendant by dialing the Operator Access Code, or to call a particular Attendant
Console by dialing an Individual Attendant Identification Number.
A-69 Automatic Idle Return
This feature enables a Dterm to become idle automatically after a predetermined time
when the line is released by the opposite Station on-hook, etc., on hands-free or the
speaker operation.
A-78 Automatic Number Identification (ANI)
This feature automatically sends the Calling Subscriber’s Number to the Called Party
after a response is received from the Distant Office. This feature is used for Outgoing
Central Office Calls [Enhanced 911 (E911) Calls] or TIE Line Calls using MF Signaling.
A-82 Automated Attendant
This feature allows a public or private network user to access the system without the
Attendant or station user’s assistance. After or during an announcement, the outside
user may originate calls over any or all of the System facilities.
A-98 Answer Hold - Attendant
This feature enables an Attendant to answer an Incoming Call by pressing the ANSWER
Key or Flashing ICI Key. If the Attendant is already engaged in a call, pressing one of
these keys places the first call on hold and automatically connects the second one. Use
of ANSWER Key or ICI Key speeds call handling, while Answer Hold prevents accidental
call dropping.
A-99 Authorization Code Display Elimination
This feature eliminates AUTHORIZATION CODE [A-20] and FORCED ACCOUNT
CODE [F-7] from being displayed on the Dterm for security purposes.
A-105 Attendant Overflow
This feature allows Incoming Trunk Calls directed to the ATTENDANT CONSOLE [A-3]
to overflow to a predetermined Night Transfer Destination Station. The Trunk Call will
ring the Attendant for a pre-programmed time period before overflowing to the
Destination Station.
This feature enables the Incoming Trunk Calls, directed to the ATTENDANT CONSOLE,
to overflow to a predetermined Station in the FCCS Network via FCCS, also.
A-121 Add-On Conference - 8 Party
This service allows the 8-Party Conference to be established any time by any Conferee
by use of the 8-Party Conference Trunk.
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A-125 Analog Caller ID (Class SM)
This feature allows the Dterm to display Calling Party’s Number and/or Name on the
LCD when a call from an analog Central Office Line is terminated. In addition, the Calling
Number can be output as a part of SMDR information. Caller ID information can be
displayed on an Analog Telephone with a display device or the display of a Dterm.
A-127 Analog Caller ID - Station
This feature allows a Called Station to receive CALLER ID information (time, Calling
Number and Name) from Central Office when a call terminates via the analog COT. The
Calling Number received from the ISDN network, MF-ANI information received from
Central Office with the MF feature or the calling Telephone Number within the the
System can change into CALLER ID information and be sent to the Called Station. This
feature is also provided for the Analog Station equipped with the function of receiving the
Calling Number. This function is called Analog Caller ID - Station by Modem Sender.
A-139 Account Code Override
This feature allows an Account Code to be entered during a conversation, overriding the
original Account Code entered for that conversation.
A-142 Anonymous Call Block
This feature allows a Station User to block calls that do not present a Calling Party
Number. Anonymous Call Block also allows a Trunk Caller to be routed to a common
announcement instead of receiving a Busy Tone.
B-2 Busy Lamp Field - Flexible
This feature provides the Attendant with visual indications of either busy, idle or LINE
LOCKOUT [L-3] conditions for a particular group of Stations via a designated lamp on
the Attendant Console or the dial pad on the Desk Console.
B-3 Busy Verification
When an Attendant places a call to a busy Station, this feature allows the Attendant to
break into the connection. When this feature is initiated from the Console, the system
sends a Warning Tone to the Stations before establishing a Three-party Conference.
Additionally, the Attendant Monitor Function can be activated depending on System Data.
B-5D Boss - Secretary Override - Dterm
This feature enables a Secretary to Voice Announce a call to a Boss when he is
currently on his My Line.
B-6 Brokerage - Hotline
This feature provides a HOTLINE [H-1] function for Dterm.
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B-8 Boss - Secretary - Message Waiting Lamp Control
This feature allows a Secretary, via the Station, to control the Call Indicator Lamp Note
or Message Waiting (MW) display on the Boss’ Station.
B-11 Boss - Secretary Transfer
This feature allows a Secretary to Voice Announce a call to a Boss when the Secretary
answers a call to the Boss’ Line.
B-12 Brokerage Hotline - Outside
This feature provides a HOTLINE - OUTSIDE [H-9] function for a Dterm.
B-17 Blind Transfer To Attendant
This feature allows a Station to transfer a Held Call (Station or Trunk) to the Attendant
Console and hang up without having to wait for the Attendant to answer.
B-18 Blind Transfer To Station
This feature allows two Service Feature Classes to be assigned to Stations, enabling
services such as Voice Mail, Announcement Machine, etc. depending on the Service
Feature Class.
1. A Service Feature Class which allows No Answer Timer to be extended when the
Blind Transferred-to Station does not answer within the predetermined period of time
interval.
2. A Service Feature Class which activates CALL FORWARDING - DON’T ANSWER [C3] set to the Blind Transferred-to Station when CALL FORWARDING - DON’T ANSWER
[C-3] is available.
B-29 BLF over FCCS
The system can exchange Busy Lamp Field (BLF) data. As shown below, the busy
status information of the other nodes within FCCS network can be displayed on the
ATTCON of the Center Node designated by the System data..
C-1 Call Back
This feature provides the ability for a Calling Station that has dialed a busy Station to dial
a Call Back code. When this has been done, the Calling Station will be rung as soon as
the busy Station becomes available, provided the Calling Station is also idle.
C-1D Call Back - Dterm
This feature provides the ability for a Dterm, which has dialed a busy Station, to press a
CALL BACK Key. When this has been done, the Calling Station will be rung as soon as
the busy Station becomes available, provided that the Calling Station is also free.
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C-2 Call Forwarding - Busy Line
This feature permits a call to a busy Station to be immediately forwarded to a
predesignated Station or to the ATTENDANT CONSOLE [A-3]. If a Called Station is in a
STATION HUNTING [S-7, 8, 9] Group and forwards calls to another Hunt Group, it can
be determined through System Data that the Calling Party has been directed either to
the Called Parties Hunt Group or to the Terminating Parties Hunt Group when all of the
Forwarded Stations are busy.
C-2D Call Forwarding - Busy Line - Dterm
This feature permits a call to a busy Station to immediately be forwarded to a
predesignated Station, or to the Attendant. CALL FORWARDING - BUSY LINE - Dterm
may be set or cancelled by the user for all Multi Line Appearances on the Dterm.
Additionally, a Single Line user may set CALL FORWARDING - BUSY LINE – Dterm for
all Sub Lines on the Dterm.
C-3 Call Forwarding - Don’t Answer
This feature permits a call to an unanswered Station to be forwarded to a predesignated
Station, or to the Attendant, when the Called Station doesn’t answer after a
predetermined time interval. If the individual Call Forwarding - Don’t Answer Timer is not
assigned, the Default Timer assigned by System Data programming is used.
C-3D Call Forwarding - Don’t Answer - Dterm
This feature permits a call to an unanswered Station to be forwarded to a predesignated
Station or to the Attendant, if the Called Station does not answer within a predetermined
period of time. CALL FORWARDING - DON’T ANSWER - Dterm may be set or
cancelled by the user for all Multi Line Appearances on the Dterm. Additionally, a single
Station user may set CALL FORWARDING - DON’T ANSWER - Dterm to all Sub Lines
on the Dterm.
C-5 Call Forwarding - All Calls
This feature permits all calls destined for a particular Station to be routed to another
Station (or to the Attendant) regardless of the busy or idle status of the Called Station.
Activation and cancellation may be accomplished by either the individual user or the
Attendant.
C-5D Call Forwarding - All Calls - Dterm
This feature permits all calls destined for a particular Station to be routed to another
Station, or to the Attendant, regardless of the busy or idle status of the Called Station.
Activation and cancellation may be accomplished by the user or the Attendant. CALL
FORWARDING - ALL CALLS - Dterm may be set or cancelled by the user for all Multi
Line Appearances on the Dterm. Additionally, a single user may set CALL
FORWARDING - ALL CALLS - Dterm to all Sub Lines on the Dterm.
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C-6 Call Hold
This feature permits any user to hold a call in progress by performing a switch hook flash
and dialing a preprogrammed CALL HOLD Code, thus allowing that Line to be used for
originating another call or returning to a previously Held Call.
C-6D Call Hold - Dterm
This feature permits a Dterm user to HOLD a call in progress by using CALL HOLD Dterm.
C-7 Call Pickup - Group
This feature permits a user to answer any calls directed to other Lines in the user’s
preset CALL PICKUP – GROUP by dialing a Pickup Code. A user can be assigned to an
additional CALL PICKUP - GROUP, referred to as EXPANDED CALL PICKUP - GROUP.
C-7D Call Pickup - Group - Dterm
This feature permits a user to answer any call directed to another line in his preset CALL
PICKUP - GROUP by using a programmable Line/Feature Key.
CALL PICKUP - GROUP - Dterm may be used by seizing Dial Tone from any Multi Line
Appearance on the Dterm. A user can be assigned to an additional CALL PICKUP GROUP, referred to as EXPANDED CALL PICKUP - GROUP.
C-8 Call Processing Indication
This feature provides visual indications at the ATTENDANT CONSOLE [A-3] of all calls
being handled by the Attendant.
C-9 Call Queueing
This feature enables an Attendant to handle a series of Exchange Network Calls in the
order of their arrival, eliminating unnecessary delays.
C-10 Call Transfer - Attendant
This feature permits a user, while connected to an Exchange Network Call, to signal the
Attendant and have the Attendant transfer the call to another Station within the system.
C-11 Call Transfer - All Calls
This feature permits a user to transfer Incoming or Outgoing Central Office and Intraoffice Calls to another Station within the system without Attendant assistance.
The user can set Camp-On function to the busy destination while transferring an
Incoming Call (Camp-On by Station). With this feature, the Trunk Party is called back
directly, as the destination becomes idle.
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C-11D Call Transfer - All Calls - Dterm
This feature permits a Dterm user to transfer Incoming or Outgoing Calls without
Attendant assistance. The Dterm user can set Camp-On function to the Busy Destination
while transferring an Incoming Call (Camp-On by Station). With this feature, the Trunk
Party is called back directly as the destination becomes idle.
C-12 Call Waiting - Terminating
This feature enables a busy Station to receive a second Incoming Call. A Camp-On
Indication Tone is sent to the busy Station, the user can use a switch hook flash to
answer the second call. A switch hook flash may be used to alternate between the two
calls.
C-13 Call Waiting Lamp
This feature provides a visual indication to the Attendant when one or more calls are
waiting to be answered.
C-14 CCSA Access
This feature enables connection to a Common Channel Switching Arrangement (CCSA)
Network. CCSA Networks provide customers with the use of a completely private interfacility dial system. They use individual facilities for dedicated Access Lines and Trunks
that terminate in Common Control Switching Equipment and in various types of
Telephone Equipment at Customer Locations.
C-15 Class of Service - Individual
This feature permits all SV7000 Stations to be assigned a Class of Service in
accordance with the degree of system use desired. Each Station is assigned one Class
from each of three groups.
C-17 Consultation Hold - All Calls
This feature permits a user to hold any Incoming or Outgoing Public Network or TIE Line
Call, or any Intra-Office Call, while originating a call to another Station within the system.
C-17D Consultation Hold - All Calls - Dterm
This feature permits a Dterm user to hold any Incoming or Outgoing Public Network or
TIE Line, or any Intra-Office Calls, while originating a call to another Station within the
system.
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C-20 Centralized Attendant Service (CAS)
For applications requiring multiple sites, but having common interests or operation, the
System can operate in a Main (attended) or Satellite (unattended) configuration.
Typically, Incoming Calls to the Satellite are routed to the Main Location for processing
by the Central Attendants within the CAS Network. This feature can be used to improve
overall communication efficiency.
C-21D Code Calling Access - Dterm
This feature provides the ATTENDANT CONSOLE [A-3] and users dial access to Code
Calling Equipment.
C-22D Called Station Status Display - Dterm
This feature provides for the status of a Called Station to appear on the LCD of a Calling
Dterm.
C-24D Calling Number Display - Dterm
This feature provides for the Station and Trunk Number of an Incoming Call to appear on
the Dterm’s LCD. This display will flash while the call is ringing, then appear steady
when the call is answered.
C-25 Call Forwarding - Intercept / Announcement
This feature provides for interception of STATION-TO-STATION [S-11], DIRECT
INWARD DIALING [D-8], Attendant-to-Station and CCSA ACCESS [C-14] Calls that
cannot be completed (unassigned Station, Level, etc.). These calls are automatically
routed to a Recorded Announcement informing the caller that an Inoperative Number
was reached and giving the Listed Directory Number for information. This feature
permits a Station-originated Call, upon encountering a restricted Outgoing Number, to
automatically be routed to a Recorded Announcement informing the caller that the
Dialed Number is restricted for this Station.
This feature permits a Station-originated Call, upon encountering a Trunk Busy
Condition, to automatically be routed to a Recorded Announcement informing the Main
System caller that all the Outgoing Trunks are busy.
C-26 Call Forwarding - Override
This feature allows a Target user (Station A) to call the Station (Station B) which has set
CALL FORWARDING - ALL CALLS [C-5] to it. If the Called Station (Station B) has set
the CALL FORWARDING - BUSY LINE [C-2] to the Calling Station (Station A), the
Calling Station (Station A) hears Busy Tone and can activate CALL BACK [C-1],
EXECUTIVE RIGHT-OF-WAY [E-1], or CALL WAITING - ORIGINATING [C-31].
C-27D Call Waiting - Answer - Dterm
This feature allows a Dterm user to answer a CAMPED-ON [A-1] or CALL WAITING [C12, 31] Call while putting an existing call on HOLD [C-6], by pressing the ANSWER Key.
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C-28 Call Forwarding - All Calls - Outside
This feature allows a call, which originates from a Station or Trunk and is destined for a
Station, to be routed to another Station outside the System via the appropriate Trunk
Line. This feature can also be activated from a Dterm Multi Line.
C-29 Call Park
This feature enables the Attendants or Station Users to “Park” calls against their own
Extension Numbers. Calls can easily be retrieved from any Station within the system.
C-30 Call Pickup - Direct
This feature allows a user to pick up a call to any other Station in the system by dialing a
specific CALL PICKUP - DIRECT code.
C-31 Call Waiting - Originating
This feature provides selected Stations with Camp-On capability to a busy Internal
Station.
C-33 Consecutive Dialing - Attendant
This feature gives the Attendant Console the ability to generate DTMF Signals while
engaged in a Station/Trunk Connection. These DTMF Signals are generated via the dial
keypad of the Attendant Console.
C-43 Call Metering
This feature enables the Metering Pulses received from an Exchange Line involved in an
established connection to be detected, counted, and stored as billing information. This
information is output to Station Message Detail Recording (SMDR), following the release
of the line.
C-59 Called Number Display - Attendant
If a call is transferred to the ATTENDANT CONSOLE [A-3] as a result of:
DO NOT DISTURB[D-11]
CALL FORWARDING - ALL CALLS[C-5]
CALL FORWARDING - BUSY LINE[C-2]
CALL FORWARDING - DON’T ANSWER[C-3]
CALL FORWARDING - INTERCEPT / ANNOUNCEMENT[C-25]
C-60 Call Forwarding - Busy Line - Outside
This feature allows a call that originates from a Station or Trunk, and is destined for a
Station to be routed to another Station outside the System via the appropriate Trunk Line.
This feature can also be activated from a Dterm Multi Line.
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C-62 Consecutive Speed Calling - System
This feature allows a user to call frequently dialed DID Numbers using fewer digits
(Abbreviated Call Codes) than would normally be required. The stored digits would
include Route Access Code, Area Code and Office Code. The caller would then
manually enter the Called Telephone Number.
C-69 Call Forwarding - All Calls - Announcement
This feature permits all calls destined for a particular Station to be routed to a Recorded
Announcement. Activation and cancellation may be accomplished by either the
individual user or the ATTENDANT CONSOLE [A-3].
C-73 Call Back - Delayed
This feature allows a Station to which CALL BACK [C-1] has been set to initiate another
call, within a predetermined time period after becoming idle and before the CALL BACK
[C-1] feature is activated.
C-74 Call Forwarding - Don’t Answer - Outside
This feature allows a call which originates from a Station or Trunk, and is destined for a
Station, to be routed to another Station outside the System via the appropriate Trunk
Line. This feature can also be activated from a Dterm Multi Line.
C-75 Call Forwarding - Intercept / Announcement - Attendant
This feature provides for interception of Attendant-originated Calls that cannot be
completed (unassigned Station, Level, etc.). These calls are automatically routed to a
Recorded Announcement informing the caller that an Inoperative Number was reached.
C-76 Call Waiting Lamp - UCD
This feature uses an LED indication, which is one of the programmable keys on the
Dterm, to indicate when a call is waiting to be answered in the UCD Queue.
C-81 Call Forwarding - I’m Here
This feature permits the CALL FORWARDING - ALL CALLS feature to be set/cancelled
from the Target Station.
C-109 Centrex Compatibility
This feature allows a DTMF Type Telephone user engaged in a Trunk Call to send a
Hooking Signal to the Called Office, enabling the call to be transferred to another Station
in that Office.
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C-133Call Forwarding - Don’t Answer To UCD Queue
This feature allows calls to a Station that has set CALL FORWARDING - DON’T
ANSWER to a UNIFORM CALL DISTRIBUTION Group to be added to a UCD Queue if
all Stations in the UCD Group are busy. (Originally, calls forwarded to a UCD Group
would receive Reorder Tone if all Stations in the group were busy.)
C-146 Call Hold - Conference
This feature allows a user to establish a Conference Connection by adding a third party
who is held on a Multi Line of a Dterm to the existing Two-party Connection. The Held
Call which is to be added may be a Station or Trunk Call.
C-150 Call Block
This feature is a part of Caller ID Service. This feature restricts termination of Incoming
Calls from the preassigned Calling Numbers. This feature is available by designating the
Calling Number by the MAT. Call Block is used to restrict an Incoming Call that has a
Restriction Number (Physical Station Number, Telephone Number or the Caller ID
Number, which are preset by each Station). The Restriction Number can be set or
cancelled at each Station by using the Access Code or the Feature Key. The Called
Station can send Busy Tone to the caller.
C-151 Call Return
This feature allows the Dterm or an Analog Station to save the received Calling Number
and the Received Number of the Station. The following two types of saving features are
available:
• Automatic Saving:The received Calling Number/Number of the Station is automatically
saved when the Station rings. The Stored Number can be dialed by pressing the Feature
Key or dialing the Access Code.
• Manual Saving:The received Calling Number is manually saved by the Dterm user
while in conversation The Stored Number can be dialed by pressing the Feature Key.
C-154 Call Hold Pick Up
This feature allows a user to pick up a Trunk or Station Call that is put on hold by a
second user. The user can accomplish this by dialing an Access Code followed by the
Telephone Number that is holding the call.
C-157 Calling Party Number - Name Assignment and Display
This feature allows a user to assign the desired name for the Calling Number, which is
provided by Central Office, CCIS or ISDN feature, using the MAT command. The
assigned name is displayed on the LCD of the Dterm and the Desk Console.
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C-160 Call Trace
This feature enables the output of System Message [26-R], that gives information on a
Malicious Call (kind of call, number of Calling Terminal, etc.) originated from a
Station/Trunk. The feature is made available when an Access Code is dialed or the Call
Trace Key is pressed by the called user.
C-169 Consultation Hold Release
This feature enables a Station, connecting to a second call while the Station has been
holding (Consultation Hold) an original call (including ADD-ON CONFERENCE - 8
PARTY [A-121] member), to disconnect the second call forcibly by pressing the DISC
Key once for Dterm, or switch hook flashing for Analog Stations/PSs twice. After that, the
second call is released, and the original call is retrieved from hold.
C-172 Call Forward by Service Feature Class
This feature will implement all types of Station Based Call Forwarding (Call Forwarding Busy Line, Call Forwarding - Don't Answer, Call Forwarding - All Calls) to a
predetermined number, based on the Service Feature Class (SFC) of the Called Station.
This feature may be set using Common or Split Forwarding, depending on system
programming.
C-173 Call Redirect
This feature allows a Dterm user to view on the Dterm display the Station Number or
Caller ID of an Incoming Call and immediately redirect the call by pressing a Function
Key. The destination of the Call Redirect will be the Call Forwarding - Don’t Answer
Destination or the Recall Destination (if the call is transferred and no Call Forwarding Don’t Answer Destination is set).
C-174 Calling Station Number Display Elimination
This feature eliminates the Calling Station Number display shown on the Dterm for
privacy or security purposes.
C-176D Call Park Group - Dterm
This feature enables the Dterm users to make a connected call into Call Park state by
pressing the Dterm HOLD Key. The Held Call can be retrieved from any Member
Stations of the Group by using pre-assigned Feature Key. When using this feature, any
of the line keys (My Line/Sub Line) is not necessary to be used for holding a call. Thus,
Station can originate/receive another call by using a line key while holding the call.
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C-177 Caller ID on Call Waiting
This feature (available since Release 11 software) allows Incoming Trunk Calls with
Caller ID information to be displayed on a Caller ID device while the user is on another
Station or Trunk Call. A standard Trunk Call Waiting Indication Tone is sent to the Busy
Station. Caller ID information for the Calling Party that is in a Call Waiting Mode is sent
to the Caller ID device. This allows the user to see the Caller ID information on the new
caller and decide if
they wish to use a switch hook flash to answer the second call. The user may also
alternate between the two calls by switch hook flash.
C-192 Called Station Name Display on CF
This feature allows the name data of a Call Forwarded Station to be displayed on the
Dterm target of the forward.
C-194 Call Park Group
This feature enables the Station Users to “Park” calls against their own Extension
Numbers. Maximum of five calls can easily be parked and retrieved from any Station
within the Call Park Deluxe Group by just pressing the access code or the pre-assigned
Dterm Line Feature key (including soft key). When retrieving a parked call from the
station out of the CALL PARK DELUXE group, station must dial the access code (or
press the Dterm Line Feature Key) +
station number of the station actually parked the call.
D-2 Dial Access To Attendant
This feature allows a user to access the ATTENDANT CONSOLE [A-3] by dialing the
Operator Call Code.
D-3 Dictation Access
This feature permits users dial access to customer provided Dictation Equipment, and in
some instances allows them to maintain telephone dial control of normal dictation
System features.
D-5 Digital Display - Station
This feature provides the Attendant with a visual display (via the ATTENDANT
CONSOLE [A-3]) of the Telephone Number, its Trunk Route Restriction, CLASS OF
SERVICE - INDIVIDUAL [C-15] and TENANT SERVICE [T-12] Number during
Attendant-to-Station Connection.
D-6 Digital Display - Trunk
This feature provides the Attendant with a visual indication of Incoming and Outgoing
Trunk Calls on the ATTENDANT CONSOLE [A-3]. Trunk Identification Number, Trunk
Route Number, and TENANT SERVICE[T- 12] Number, or Central Office Trunk Code,
are displayed in a numerical readout.
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D-7 Direct-In Termination (DIT)
This feature automatically routes Incoming Public Network Calls directly to a Preselected Station without Attendant assistance. The call can then be processed by the
Called Party. THREE-WAY CALLING [T-2], CALL TRANSFER [C-10], etc., are handled
in the same manner as any Normal Trunk Call.
D-8 Direct Inward Dialing (DID)
This feature provides for all Incoming Calls from the Public Network (except FX or
WATS) to reach any System Station without Attendant assistance.
D-9 Direct Outward Dialing (DOD)
This feature permits any user to gain access to the Public Network without the
assistance of the Attendant, by dialing an Access Code and receiving a Second Dial
Tone. The user may then proceed to dial the desired Public Network
Number.
D-10 Distinctive Ringing
This feature provides Distinctive Station ringing patterns so that the user can distinguish
between internal and External Incoming Calls.
D-11D Do Not Disturb - Dterm
This feature allows a user to establish DO NOT DISTURB status. Incoming Calls will be
denied access to My Line, while DO NOT DISTURB status is in effect.
D-13D Dial Monitor - Dterm
This feature provides for dialed digits to appear on the Dterm LCD.
D-15 Day/Night Class of Service
This feature permits any Station to be assigned one CLASS OF SERVICE - INDIVIDUAL
[C-15], for Day and another for Night. System Data can be programmed so that once the
ATTENDANT CONSOLE [A-3] has entered the NIGHT CONNECTION [N-1, 2] mode, a
Station’s CLASS OF SERVICE - INDIVIDUAL [C-15] will be automatically changed when
required. Class may be upgraded or downgraded, and Trunk Groups normally controlled
by the Attendant in the Daytime may be opened to Station control in the Night mode.
D-16 Direct Digital Interface
This feature provides the capability to connect Trunks from the SV7000 directly to T1
Carrier Links via either Private or Public Network.
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D-31 Delay Announcement - UCD
This feature provides recorded Announcement Service for an Incoming Call directed to a
busy UNIFORM CALL DISTRIBUTION (UCD) [U-1] Group and placed in a Waiting
Queue. The Recorded Announcement occurs after Ringback Tone is sent for a
predetermined time.
D-32D Dual Hold - Dterm
This feature allows two Dterms to simultaneously be placed on hold. This allows the
Held Parties to answer or originate a call from a Sub Line Appearance or the idle Prime
Line.
D-87 Dial Intercom
This feature gives multiple Dterm users the ability to simultaneously call any other
member of the Intercom Group, regardless of their idle or busy status. An Intercom
Group member can override another Intercom Station in a Twoparty Intercom
Connection. This service is referred to as an Intercom Bridge.
D-90D Delayed Ringing - Dterm
This feature allows a Dterm to receive an Incoming Ringing Signal after a predetermined
time.
D-114 Delay Announcement - Attendant
This feature provides a Recorded Announcement to the Incoming Trunk Caller waiting to
be answered by the Attendant. This feature is divided into the following two types,
depending on the time when the call is connected with the Announcement Trunk.
1. The call is connected to Announcement Trunk immediately after terminating to the
Attendant Console. The announcement continues until the Attendant answers, or the
Trunk side abandons the call.
2. The call is connected to the Announcement Trunk after the Ringback Tone
Connection.
D-149 Direct Station Selection (DSS) Console
A Direct Station Selection (DSS) Console may be connected to each Dterm. The DSS
Keys can be used as Line Keys or Feature Keys.
D-153 Distinctive Ringing - Caller ID
This feature is a part of CALLER ID Service. This feature provides a Distinctive Ringing
Pattern (0, 1, 5, 6 or 7) according to the received Calling Number.
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D-156 Dual Station Call
This feature allows a user to simultaneously call two Stations (referred to as a Main
Station and Sub Station), which are accommodated in two different Nodes within an
FCCS Network, by dialing one Telephone Number of the Main Station. At that time, the
call can be answered from either the Main or the Sub Station. Additionally, when this
service is used with BOSS - SECRETARY Service, the Main Station can forward a call
automatically from a
Secretary Station. This service requires the data setting of FCCS. For more information
on FCCS Data, see the Data Programming Guide (Windows) - FCCS.
D-157 Dialed Number Primary Restriction
This feature (available since Release 10 software) provides the ability to block Dialed
Numbers up to 24 digits long. This function is very similar to Primary Code Restriction
via APCR. The main difference is that the number of restrictions has been expanded.
D-158 Digital Announcement Trunk Recording via Remote Access
(Series 8400 software feature)
This feature allows a user to record/playback an announcement on 4DAT via remote
access to the System using an external Central Office Trunk. This is required by any
user who needs to re-record or to replay (playback) DAT Announcements because of
severe weather closure of Office/Call Center. This feature provides the following four
functions:
•Recording/Playback by designating Route and Trunk
•Recording/Playback by designating Route
•Recording/Playback of an announcement for a UCD Group
•Recording/Playback of an announcement for ACD
D-159 DtermIP
This feature allows the system to use the following two Voice Over IP-capable
telephones, Dterm IP and DtermIP INASET, as a Station.
D-160D Dynamic Dial Pad - Dterm
This feature allows a user to make an Outgoing Call by dialing the desired Station
Number (Digit Keys 0 - 9, *, and #) without pressing Speaker Key or going Off-Hook.
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D-161 Dterm Assistant
Dterm Assistant is a maintenance application software installed in Web server, which is
located between System (device) and Web client on a network.
Dterm Assistant enables the clients to access the server and assign the terminal data
such as feature key arrangement on Dterm via Web browser (Internet Explorer) using
their own PC. It is not necessary for the user to install any special application software
other than Internet Explorer.
Due to these advantages, parts of the Office data settings get to be assigned by the user
without asking the System maintenance personnel to operate conventional MAT
command.
User ID of clients must be assigned to log in to Dterm Assistant server. As the clients are
categorized by user class, there is certain restriction of service utilized depending on the
user class for acquiring security measures.
D-162 Distinctive Ringing List
(Series 8800 software feature)
This feature allows a Station the ability to set 16 individual CPNs in bins, up to 16 digits
each. These numbers will then ring the station with a Distinctive Ringing Pattern.
D-163 Distinctive Ringing for Toll Calls
This feature (available since Release 12 software) allows Incoming Calls from outside of
a defined Local Area to ring with Ringer Pattern 5, while Incoming Calls from the defined
Local Area will ring with Ringer Pattern 1.
D-164D Day-Night Mode Status - Dterm
This feature allows a Dterm telephone the ability to display the current System status
and change the System into Day or Night using either the “*” or “#” key.
D-166 Dial Plan to support IP Centrex Functionality
This feature is useful when changing an existing network to IP-based network. In the
following example, two switching systems installed in branch offices are removed.
Telephones in the branch office are accommodated in the Head Office. And then the
users can use the existing telephone services without their own switching system
(Centrex Functionality). Thus this feature can bring about the cost reduction of
telecommunications. In addition, the telephone users can use the previous numbering
plan to which they have been accustomed - Dial Plan is supported.
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D-165 Dialed Number Privacy Mode
This feature allows the Dterm user to change the Dialed Number Privacy mode at any
time. The following three modes are available:
Normal Mode: Dialed Number is displayed on the LCD of Dterm as usual.
In this mode, Last Number Called - Dterm [L-6D] is available.
Privacy Mode 1: The Secret Code (the default is “*”) is displayed on the LCD of
Dterm instead of the dialed number.
In this mode, Last Number Called - Dterm [L-6D] is not available.
Privacy Mode 2: The Secret Code (the default is “#”) is displayed on the LCD of
Dterm instead of the dialed number.
In this mode, Last Number Called - Dterm [L-6D] is available.
E-1 Executive Right-of-Way
This feature enables selected users, upon encountering a busy condition at an Internal
Station, to bridge into the Busy Connection.
E-1D Executive Right-of-Way - Dterm
This feature enables selected Dterm users, upon encountering a busy condition at an
Internal Station, to bridge into the Busy Connection after transmitting a Warning Tone.
E-3D Elapsed Time Display - Dterm
This feature provides an LCD display of the time elapsed while a Dterm is connected to
any Trunk.
E-4D Exclusive Hold - Dterm
This feature allows a user to place a call on hold and to exclude all other users from
retrieving the Held Call.
E-11 Emergency Call
This feature allows a user to make an EMERGENCY CALL by dialing a Special
Telephone Number (Emergency Telephone Number). Even if the Calling Station goes
on-hook, the Calling Station is recalled and the speech path is maintained until the
Emergency Telephone goes on-hook. When a Station dials the Emergency Telephone
Number, a System Message is printed out to report the EMERGENCY CALL.
E-18 E911-ANI Unified Number of Digits
This service unifies the number of ANI information digits sent out by E911-ANI Service.
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E-22 Emergency System Control
This feature provides the emergency terminating method to Attendant Console and the
originating call restriction. Operator calls which are originated from Local nodes
belonging to a Hub node during an emergency mode terminate to VMS, IVR or the
specific ATTCON in the Main node, and the operator there will answer and operate. All
outgoing C.O. calls originated during an emergency mode are restricted from Local
nodes belonging to a Hub node.
E-26D Expanded Multiple Line Operation - Dterm
This feature allows operation of whichever lines from ELC Cards in the same IMG or in
different IMGs in the same Node/LMG, as a Dterm’s programmable Multiple Lines (My
Line + Prime Line/Sub Lines). While Multiple Line Operation - Dterm [M-20] only allows
such a programming within the reach of ELC Cards in the same IMG, this feature
enables the Multiple Line Operation beyond the reach of a single IMG (though the ELCs
cannot be in different Nodes/LMGs).
E-27 Expanded Individual Speed Dial
This feature allows a user to dial certain frequently called exchange network numbers
using fewer digits (abbreviated call codes) than normally required. Using this feature, a
user can establish his own abbreviated codes.
E-28 Expanded Individual Speed Dial - Group
This feature This feature (available since Release 13 software) allows a user to share a
set of common Speed Dial numbers with other users in the group.
E-29 Expanded Account Code
This feature is an adjunct to STATION MESSAGE DETAIL RECORDING [S-10], which
provides a station user with the capability to enter a cost-accounting or client-billing code
(up to 24 digits) into the system before dialing a long distance number.
F-1 Flexible Numbering of Stations
This feature provides the ability to assign Telephone Number of the Voice Station and
Data Station to any corresponding instrument location, depending solely upon
Numbering Plan limitations.
F-2 FX Access
This feature provides dial access to distant Central Offices using Foreign Exchange (FX)
Trunks. All Incoming Calls to the System from the FX Central Office area are placed to
the listed Foreign Exchange Directory Number and are answered by the Attendant in a
manner similar to LDN Service. Outgoing Calls can be made on an Attendant handled
basis using direct or dial access and/or on a direct basis by Station.
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F-3D Flash Button - Dterm
This feature provides for a Dterm set to program one of the programmable Line/Feature
Keys to function as a FLASH BUTTON. The FLASH BUTTON will function in the manner
a switch hook flash does for a Single Line Set.
F-4D Flash Entry - Dterm
This feature allows a Dterm user to insert a switch hook flash as the first digit in a stored
SPEED CALLING – ONE TOUCH - Dterm [S-26D] Number.
F-5D Flexible Ringing Assignment - Dterm
This feature enables Dterm Line Appearances to be individually programmed for ringing,
not ringing, ringing (Day only), or ringing (Night only).
F-6 Faulty Trunk Report
This feature allows a Station to report a noisy or faulty Trunk Number by dialing a
Special Access Code before hanging up. The FAULTY TRUNK REPORT consists of a
Trunk Number, Telephone Number, Associated Time Division Switch and Reported
Time. This information is displayed at the MAINTENANCE ADMINISTRATION
TERMINAL (MAT) [M-18] and/or system printer.
F-7 Forced Account Code
This feature is a variation of the AUTHORIZATION CODE [A-20] feature, which makes it
mandatory to enter an ACCOUNT CODE (up to ten digits) for all Outgoing Calls. The
Account Code must be dialed before dialing the Outgoing Number. Calls are processed
only when the dialed Account Code is valid. The FORCED ACCOUNT CODE can be
masked on the Dterm display if necessary using the AUTHORIZATION CODE DISPLAY
ELIMINATION [A-99].
F-17 Flexible Assignment of Function Buttons
This feature allows Dterm Feature Buttons (Keys) (REDIAL, FEATURE, SPEAKER,
CONF, ANSWER, RECALL, HOLD, and TRANSFER) to be flexibly assigned on each
individual Dterm.
Feature Buttons are programmed through the Maintenance Administration Terminal
(MAT) and are part of System Data.
F-31 Follow Phone
This feature allows individual Station Data of a Terminal to be exchanged with another,
without the assistance of the MAT. The result of the exchange is printed out in the
System Message. The exchange of Terminal Data by this service is called SWAP
Service. The following information is included in the System Message:
a.) Successful activation:Station Data Change Notice.
b.) Unsuccessful activation:Station Data Change Notice and Error Code to indicate the
restricted
condition.
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F-35 Flexible Routing - FCCS
With this feature, an FCCS Trunk Call can automatically be routed via a Non-FCCS
Trunk (Central Office or TIE Line), if all the FCCS Trunks are busy or a Data Link Failure
occurs at the FCH Card.
F-36 FCCS Networking via IP
This feature allows the system to exchange both Speech and FCCS Signals over an
Intranet. This feature can be activated by using Internal PHF. Note The Speech data
between the IP Stations can be sent/received in a Peer-to- Peer connection, while the
control signals are handled by the System. The following are typical connection patterns
for this feature.
F-37 FCCS Alternate Routing - Backup
This feature automatically routes a FCCS over IP call over Alternate Routes when the
first-choice link is down due to the LAN failure, etc.
G-2 Group Calling
This feature allows a user to dial certain frequently Called Stations within a System
Group using an easily remembered number. The same number can be assigned to
different user groups. The Group Call can initiate CALL
G-3 Ground Button Access
This feature designates the GROUND BUTTON provided on a Station Telephone as
either the only means or an optional means of performing any available Hooking
Services.
H-1 Hotline
This feature permits a pair of Station sets to be associated with one another on an
automatic ringdown basis.
H-2 House Phone
This feature allows selected Stations to reach the Attendant Console by going off-hook.
H-4D Hands-free Answer Back - Dterm
This feature allows the user to respond to a VOICE CALL - Dterm [V-2D] without lifting
the handset.
H-5D Hands-free Dialing/Monitoring - Dterm
This feature allows the Dterm user to dial or monitor and call without lifting the handset.
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H-9 Hotline - Outside
This feature allows a user to access an outside destination by going off-hook or selecting
the HOTLINE – OUTSIDE [H-1] Line/Feature Key and automatically dialing a SPEED
CALLING - SYSTEM [S-3] Number.
H-24 H.323 Terminal Support
This feature allows the SV7000 to communicate with Gate Keeper (“GK 1000”
manufactured by NEC) via PHE. By using this feature, H.323 Terminal controlled by
Gate Keeper can be used as SV7000 Station. At this time, the voice data between GKcontrolled H.323 Terminal and SV7000-controlled IP Terminal such as Dterm IP, IP
Enabled Dterm, IP PAD, Analog MC, IPBS can be sent/received in Peer-to-Peer
Connection.
I-1 Immediate Ringing
This feature causes Called Stations to ring immediately upon establishment of
connections.
I-2 Incoming Call Identification
This feature allows an ATTENDANT CONSOLE [A-3] to visually identify the type of
service and/or Trunk Group that is arriving or waiting to be answered.
I-3 Incoming Central Office Call to TIE-Line Connection
This feature permits an Attendant to connect an Incoming Public Network Exchange Call,
via a TIE Line, to a Station at a distant office.
I-4 Individual Trunk Access
Individual Trunk Access allows Station/Attendant Console users to seize specific Trunk
by dialing. By using the feature, the user can perform the Connection Tests. For details,
refer to table below:
I-5 Inter Position Transfer
This feature allows Attendants to transfer calls at their ATTENDANT CONSOLE [A-3] to
another Attendant’s Console in systems where MULTIPLE CONSOLE OPERATION [M4] has been provided.
I-6 Individual Attendant Access
This feature permits a user to call a particular ATTENDANT CONSOLE [A-3] via an
Individual Attendant Identification Number.
I-7D I-Hold Indication - Dterm
This feature provides the Dterm with a distinctive flash to differentiate between a call the
user placed on hold from other calls.
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I-8D I-Use Indication - Dterm
This feature provides the Dterm with a unique LED indication to display the particular
line the user is on.
I-9 Intercom Calling
This feature allows a Sub Line Appearance, assigned to a Dterm, to be used for
INTERCOM CALLING.
I-10D Intermediate Station Number Display - Dterm
This feature provides an LCD to the Called and Calling Dterm Station when an
Intermediate Station has been involved in call processing.
I-11 Inter-Office Off-Hook Queuing
This feature can be employed in a Main-Satellite configuration to allow a Satellite user to
queue for LEAST COST ROUTING - 3/6-DIGIT [L-5] at the Main Location. This feature
can also be used when all Outgoing Facilities are concentrated at the Main Location.
I-20 Immediate Ringback Tone
IMMEDIATE RINGBACK TONE will be heard by the caller immediately upon
determination by the system that the Called Station is idle. IMMEDIATE RINGBACK
TONE is connected, even if the call occurs during the OFF portion of the ringing cycle.
I-21 Internal Zone Paging
This feature allows an individual to make a voice call through the Dterm speakers of the
designated group by dialing the INTERNAL ZONE PAGING Access Code or pressing
the INTERNAL ZONE PAGING Key on the Dterm.
I-43 IP Enabled Dterm
This feature allows the system to use the Dterm Series i, Dterm Series E/Dterm75
telephone sets on the LAN. A Dterm Series i, Dterm Series E/Dterm75 telephone set
(hereafter referred to as IP Enabled Dterm) that is equipped with an adapter, can send
or receive Voice Signals to or from other IP terminals (such as an Analog MC, other IP
Enabled Dterm, or IP PAD Circuit Card). An IP Enabled Dterm can be relocated by
simple plug-in operation, without changing
the office data. At this time, the control signals are processed by the System via Protocol
Handler for IP Enabled Dterm (PHD), while the speech path is established directly
between the IP terminals (voice data are sent/received in a Peer-to-Peer connection).
I-44 IP Control-FAX
This feature allows the user to specify jitter-buffer size, payload type and payload size
for FAX connection via selfoffice IP PAD and Analog MC, from the MAT.
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I-45 Individual Call Hold Recall Timer
This feature (available since Release 11 software) provides the Dterm with the ability to
individually set the options for the Call Hold Recall Timer setting.
I-46 Individual Call Block
(Series 8800 software feature)
This feature allows up to 1023 LENS per IMG the ability to set 16 individual Call Block
bins (up to 16 digits each). Individual Call Block can be turned off or on for all Call Block
Numbers by means of an Access Code, without the need to delete the Call Block bins.
Individual Call Block allows a Trunk Caller to be routed to a common announcement
instead of a Busy Tone.
In addition, the last number received feature allows the last calling number received to
be saved without dialing the Block Number or Individual Calling Number (only one
number can be saved in this manner).
I-47 IP Device Firmware Remote Download
This feature allows IP terminals to update their firmware by downloading new firmware
file. At this time, the downloading can be performed manually at any time or
automatically at a predetermined time.
This feature is activated through the following steps.
1. The system orders the terminal to download the new firmware.
2. The new firmware in the FTP server is downloaded to the assigned terminal
dependent on the parameters defined via the MAT.
3. The terminal informs the system of download state.
4. The system requires the terminal to reboot.
5. The terminal is rebooted and registered to the system with new firmware.
I-48 IP Terminal Failover - Alternate System
For an IP Telephony System having more than one SV7000, if one of the SV7000s is
failed, IP Terminals accommodated in the failed SV7000 (called Primary SV7000) can
be backed up by the other SV7000 (called Secondary SV7000).
I-49 Internal Zone Paging - Multi-Zone
This feature allows an individual to make a Voice Call through the Dterm speakers of the
first 4 (1-4) designated IZP groups by dialing the MULTI INTERNAL ZONE PAGING
Access Code or pressing the MULTI INTERNAL ZONE PAGING Key on the Dterm.
Refer to the Internal Zone Paging [I-21].
L-1 Lamp Check
This feature allows the Attendant, by pressing the LCHK Key, to light all lamps and
sound the ATTENDANT CONSOLE [A-3] buzzer.
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L-3 Line Lockout
This feature provides for the automatic release of a Station from the common equipment
when it has remained offhook for longer than the usual interval before dialing. The
System may be programmed to return a Howler Tone to the Station in the LINE Lockout
mode.
L-5 Least Cost Routing - 3/6-Digit
This feature allows the System to be programmed to route Outgoing Calls over the most
economical facility (WATS, FX, DDD). Based on the Area Code and Office Code dialed
(six-digit analyzing), the system examines the programming Tables and chooses the
facilities in the order specified. Least Cost Routing (LCR) may also be performed on a
Sender basis (LCRS).
L-6 Last Number Called - Single-Line Station
When a Station User originates a Station-to-Station Call or an Outgoing Trunk Call, but
the call is not established, this feature allows the Calling Station to recall the same
destination by dialing only a Special Code instead of dialing all the digits of the number.
L-6D Last Number Called - Dterm
This feature allows the user of a Dterm to store the last five numbers dialed, and to
redial the numbers by pressing the REDIAL Key. The Dterm user can choose any
destination out of the last five calls that have been stored in memory, thus enabling the
user to place a call without redialing the full number.
L-7D Line Reconnect - Same-Line - Dterm
This feature allows a user to disconnect a call and receive Dial Tone by pressing the
RECALL Key.
L-8D Line Reconnect - Other-Line - Dterm
This feature allows a user to select another Line Key without going on-hook.
L-9D Line Pre-selection - Dterm
This feature provides the Station user with the ability to select an idle or ringing Line
before going off-hook.
L-10 LCR - Time Of Day Routing
This feature provides automatic routing of Outgoing Calls over alternative customer
facilities, based on the Destination Code. The system will select the most economical
route available at the time of Connection. The pattern of alternate routing can be
changed up to 8 times per day, based on a prearranged time schedule.
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L-11 Line Load Control
This feature allows the Attendant to deny a preselected group of Stations the ability to
originate calls by activating a Key. The sole intention of this feature is to temporarily
restrict the least important Internal Stations from Outward Calling during an excessive
traffic load or emergency conditions.
L-20 Line Load Control - Automatic
This feature allows the system to automatically deny a preselected group of Stations the
ability to originate calls. The sole intention of this feature is to temporarily restrict the
least important internal Stations from Outward Calling during an excessive traffic load or
emergency conditions.
L-21 Line Load Control - MAT
This feature allows the MAT [M-18] to deny a preselected group of Stations the ability to
originate calls by activating a command Key. The sole intention of this feature is to
temporarily restrict the least important internal Stations from Outward Calling during an
excessive traffic load or emergency conditions.
L-22 Line Fault Detection
This feature detects faulty conditions on exchange lines (Line Disconnection etc.), and
indicates them as “Line Faults”, restricting further seizure of the faulty Line by other
Connections.
L-24 Listed Directory Number
This feature allows Listed Directory Number Display on the ATTENDANT CONSOLE [A3] when the Attendant has answered a Listed Directory Number Call. In addition, this
feature enables ATTENDANT CONSOLE to receive a Listed Directory Number (LDN)
Call, terminated at a Remote Node, via FCCS link.
L-28 LDN Night Connection
This feature routes LISTED DIRECTORY NUMBER (LDN) Calls to a preselected Station
when the system is in night mode. In addition, this feature enables ATTCON to forward
terminated LDN Calls to preprogrammed Station via FCCS link in the FCCS Network,
also.
L-30 LDN Night Connection - Outside
This feature routes Listed DIRECTORY NUMBER (LDN) Calls to a preselected Station
Outside the system when the system is in Night mode. In addition, this feature enables
ATTCON to transmit terminated LDN Call to Outside using Outgoing Trunk in the FCCS
Network.
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L-53 Logged Out IP Station - Call Destination
This feature permits a call to an IP Station in logout state to be immediately forwarded to
a predetermined Station, Attendant Console or Announcement Equipment. The
forwarding destination can be set at each terminal in advance.
M-1 Meet-Me Paging
This feature allows a user dial access to paging equipment, and connects both parties
automatically, after the Called Party has answered the page.
M-2 Miscellaneous Trunk Access
This feature provides access to all types of external and customer provided
equipment/facilities, such as FX [F-2], WATS [W-1], CCSA [C-14], TIE LINE [T-3], and
Exchange Network, along with DICTATION [D-3], PAGING [P-1], and CODE CALLING
[C-21].
M-3 Miscellaneous Trunk Restriction
This feature provides for certain Stations and certain dial-repeating TIE Trunks to be
denied access to particular Trunk Groups, such as FX [F-2], WATS [W-1], CCSA [C-14],
TIE LINE [T-3], Exchange Network, DICTATION [D-3] or PAGING [P-1].
M-4 Multiple Console Operation
This feature allows one or more ATTENDANT CONSOLEs [A-3] to operate within the
same system.
M-7 Music On Hold
This feature allows a party to hear music while in the CALL HOLD [C-6], CALL
TRANSFER [C-10, 11], or ATTENDANT CAMP-ON WITH TONE INDICATION [A-1]
conditions. Two types of MUSIC-ON-HOLD (MOH) exists; internal MOH and external
MOH.
M-11 Meet-Me Paging - Attendant
This feature allows an Attendant to hold an Incoming Call, page the Called Party, and
connect the two after the Called Party has answered the page.
M-15 Maintenance Printout
The System can provide a hard copy of real-time System Message via the System
Printer.
M-18 Maintenance Administration Terminal (MAT)
This feature provides a man-machine interface, using a personal computer, to
accomplish such items as on-line system programming of Station and Trunk data, traffic
information, fault condition analysis, and testing of operating programs.
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M-19 Message Reminder
This feature allows a Single Line Telephone to leave a MESSAGE REMINDER at the
called Dterm set.
M-19D Message Reminder - Dterm
This feature allows a Dterm to leave a MESSAGE REMINDER at another Dterm.
M-20D Multiple Line Operation - Dterm
This feature allows for the appearance of Multiple Lines on the programmable
Line/Feature Keys of the Dterm set.
M-23 Message Waiting Lamp Setting - Attendant
This feature permits an ATTENDANT CONSOLE [A-3] to set or cancel a Message
Waiting indication on a Station provided with this feature.
M-24 Multiple Call Forwarding - Busy Line
This feature permits a call to a busy Station to be forwarded, a maximum of five times, to
preprogrammed Idle Stations.
M-25 Multiple Call Forwarding - Don’t Answer
This feature permits a call to an unanswered Station to be forwarded, multiple times, to
preprogrammed Idle Stations that do not have Call Forwarding set.
M-26 Message Center Interface
This feature provides a socket interface (LAN interface) to an external CPU for Message
Center information when a specific UCD [U-1] Group or ATTENDANT CONSOLE [A-3]
is called. This interface allows external control of Call Indicator Lamp.
M-30 Message Waiting Lamp Setting - Station
This feature allows a Station to set/cancel a Message Waiting indication or lamp. Any
Single Line Telephone or Dterm can set/cancel Message Waiting indication to any
Single Line Telephone equipped with a 90v neon lamp, or a Dterm.
M-44 Multiple Call Forwarding - All Calls
This feature permits a call to a CALL FORWARDING - ALL CALLS Station to be
forwarded multiple times to a predesignated Idle Station.
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M-47 Manual Signaling
This feature permits Dterm users to send a one-second ring to a predetermined Station.
The manual intercom SIGNAL Key (SIG Key) is operational at all times. An Intercom Call
is not required. If the signalled Station is ringing from another call, the manual intercom
signal will interrupt that ringing.
M-48 Multi Channel Recording - DAT
This feature allows a Station user to record a message to multiple Digital Announcement
Trunks (DATs) in a single operation. The message recorded in a Trunk is automatically
copied to other Trunks one after another.
M-84 MF Signaling - DID
In the Associated Channel Interoffice Signaling system, this feature allows use of
selective codes of Multifrequency (MF) signals.
MF - (ANSI) codes: A total of 15 codes are available by combining two out of six
frequencies. (ANSI)
M-85D Message Waiting Indication - Dterm
This feature allows Message Waiting (MW) information to be displayed on the middle
line (24 digits) of the LCD.
M-101 MF ANI to SMDR
This feature allows an office receiving ANI information by means of MF signal to output
the information to the expanded SMDR area.
M-105 Multiple Music on Hold Source
This feature is programmed to let the system use the so-called “Voice Prompt” functions
instead of the audible tone generally supplied from the Digital Tone Generator (DTG) or
PZ-M661. When the system is equipped with an Announcement Trunk, such as DAT,
COT or EMT, the user can activate this alternative tone provision service on condition
that the announcement port and Speech Path Memory (SPM) in the following LENs are
concatenated by nailed-down Connections.
[Locations of Speech Path Memory]
MG = 00, Unit = 2, Group = 24, Level = 0~7
MG = 00, Unit = 2, Group = 25, Level = 0~7
MG = 00, Unit = 0, Group = 24, Level = 0~7
MG = 00, Unit = 0, Group = 25, Level = 0~7
MG = 01, Unit = 2, Group = 24, Level = 0~5
MG = 01, Unit = 2, Group = 25, Level = 0~7
MG = 01, Unit = 0, Group = 24, Level = 0~7
MG = 01, Unit = 0, Group = 25, Level = 0~7
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M-107 Message Audible Alert
Messages (like voice mail or message waiting lamp control) can be realized by MCI or
digital voice integration. This feature with MCI is an adjunct to MESSAGE CENTER
INTERFACE [M-26] and MESSAGE CENTER INTERFACE - CCIS [M-67], which
provides normal voice mail service. When MCI or digital voice integration sends a new
voice mail notification to a station, a chime (audible alert) information will also sent to the
station.
Also, this feature with MCI lamp control on control is an adjunct to MESSAGE WAITING
LAMP ON - ATTENDANT [M-23] and MESSAGE WAITING LAMP ON - STATION [M30]. When a lamp on notification is sent to a station, a chime (audible alert) information
will also send to the station. The station will have a light flash followed a chime after
receiving the notification.
N-1 Night Connection - Fixed
This feature routes calls normally directed to the ATTENDANT CONSOLE [A-3] to a
preselected common Station within the system when the Night Mode has been entered.
N-2 Night Connection - Flexible
This feature provides arrangements to Route Calls, usually directed to the ATTENDANT
CONSOLE [A-3], instead to a preselected Station on a flexible, assignable basis within
the system, when the Night Mode has been entered.
N-3 Non-delay Operation
This feature allows the ATTENDANT CONSOLE [A-3] to place any Calling Party on hold,
dialing the call, and connecting the Calling and Called Parties.
N-7D Non-exclusive Hold - Dterm
This feature allows a Dterm user to place a call on Hold, from which it can be retrieved
by any Dterm Station that displays the Held Line.
N-8D Non-square Line Assignment - Dterm
This feature allows the line function buttons on the Dterm set to be freely assigned as
Line Keys or as Feature Keys. These assignments are done on a per-Station basis.
N-17 Night Connection Outside - System
This is a night transfer service on a system basis, enabling a ring-down or LDN Call to
be transferred to a preselected Station Outside the System when night mode has been
set.
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N-28 Name Display - System
This feature allows a Dterm with LCD display to indicate the user information
corresponding to the Calling Telephone Number while engaged in STATION-TOSTATION CALLING [S-11].
O-1 On-line Maintenance
This feature permits maintenance routines to be performed on the System without
affecting normal system operation.
O-2 Outgoing Trunk Queuing
This feature allows a Station user to dial a specified Access Code and enter a first-in,
first-out queue, when encountering a Trunk busy signal. As soon as an OUTGOING
TRUNK becomes available, Stations in the queue will be called back on a first-come,
first-served basis.
O-2D Outgoing Trunk Queuing - Dterm
This service allows a Dterm Station user, upon encountering a Trunk busy signal, to
enter a first-in, first-out queue.
O-6 Off-Hook Alarm
This feature allows a Station user to call the Attendant or a predetermined Station by
simply staying OFF-HOOK. The Calling Number is automatically displayed at the
ATTENDANT CONSOLE [A-3] or a Dterm series with a display.
O-7 Off-Hook Queuing
This service allows a Station user, upon encountering a Trunk-busy condition, to remain
off-hook and automatically enter a first-in, first-out queue. As soon as an Outgoing Trunk
becomes available, the switch connects the next call to this Trunk.
O-13 Overflow - UCD
1. When a call has terminated to UCD [U-1] Group A, and the Incoming Call has
encountered all Stations busy in Group A, the call is transferred to UCD [U-1] Group B, if
Group B is registered as the OVERFLOW-UCD destination.
2. If all Stations are busy in Group B, then the call is placed in queue for the originally
called UCD [U-1] Group (Group A).
O-16 Outgoing Trunk Queuing - Attendant
This feature allows an ATTENDANT CONSOLE [A-3], upon encountering a Trunk busy
condition, to remain Off- Hook and automatically enter a first-in, first-out queue. As soon
as an Outgoing Trunk becomes available, the switch connects the next call to that Trunk.
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O-21 Outgoing Trunk Busy Announcements
This feature permits a Station-originated call, upon encountering a Trunk busy signal, to
be automatically routed to
a recorded announcement informing the caller that all Outgoing Trunks are busy.
O-37D Off-Hook Line Number Display - Dterm
This feature permits the Dterm LCD to display the Calling Telephone Number/Trunk
Number when the Dterm user originates a call by pressing a Line Key or using TRUNK
LINE APPEARANCE [T-23] service.
P-1 Paging Access
This feature provides both ATTENDANT CONSOLE [A-3] and Station users dial access
to PAGING equipment. ATTCON and Station can page anyone over the speaker
connected to MC&MG-COT equipment.
P-2 Passing Dial Tone
This feature allows the Attendant to PASS DIAL TONE to a restricted Station user,
enabling that user to place a call that would normally be restricted.
P-3 Power Failure Transfer
This feature provides for certain specified Trunks to be automatically connected to
designated Stations in the event of a loss of AC power.
P-4 Pushbutton Calling
This feature permits Station users to originate calls using Pushbutton Telephones and
also allows Dterm Stations the ability to control external devices requiring DTMF Signals
to initiate or perform functions (e.g., Code-A-Phone, Conference Unit, etc.).
P-5 Pushbutton Calling - Attendant Only
This feature permits an Attendant to place all calls over DTMF Signaling lines from a
Pushbutton Keypad on the ATTENDANT CONSOLE [A-3].
P-6 Pushbutton to Rotary Conversion
This feature allows DTMF telephones to be used when DTMF Signaling is not provided
or are not available from the Central Office and/or TIE Line.
P-7 Peg Count
This feature permits traffic studies and traffic analysis information to be accessed from
the MAINTENANCE ADMINISTRATION TERMINAL (MAT) [M-18] and to be printed out.
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P-9 Priority Call
This feature allows the ATTENDANT CONSOLE [A-3] to answer various types of calls in
the order of their priority. A special type of call can be handled prior to a regular call, at
the Attendant's discretion.
P-10 Paging Transfer
This feature allows a Station user to transfer a Paged Call to a party that has been
placed on hold. Station can page anyone over the speaker connected to MC&MG-COT
equipment after holding a station/trunk call to transfer.
P-11 Periodic Time Indication Tone
This feature provides a Tone every 180 seconds, if required, to the Station user who has
made an Outgoing Call.
P-13D Prime Line Pickup - Dterm
This feature allows a Dterm Station user to originate or answer a call from the line
designated as the Primary Line by going off-hook. It is unnecessary to press the
associated Line Key.
P-14D Privacy - Dterm
This feature allows a Dterm to establish privacy so that no Station can interrupt a call via
EXECUTIVE RIGHT-OFWAY [E-1].
P-15D Privacy On All Lines - Dterm
This feature restricts Dterm users from pressing a Busy Line Button and entering a
conversation, which can occur with 1A2 Key Telephones without Exclusion Circuits.
P-18 Privacy Release
This feature allows multiple Stations (maximum of six) accommodated in the same Multi
Line Group to override a Dterm already engaged in communication. Also, by Key
operation of the Dterm already engaged in communication (maximum of 8-Party
Communication), overriding from other Stations in the same Multi Line Group can be
restricted.
P-30 Priority Paging
This feature allows a Station or an ATTENDANT CONSOLE [A-3] to make a PRIORITY
PAGING [P-30] access when a Paging Call cannot normally be originated because
another Paging Call is in progress or is waiting for an answer. ATTCON and Station can
page anyone over the speaker connected to MC&MG-COT equipment, overriding other
party’s paging in progress.
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P-57 Pad Lock
This feature temporarily restricts telephones from making unauthorized calls by dialing
special Access Code when Station users are away from their seats. In addition, the
Station user can assign the desired Route Restriction Class (RSC) from the Station.
P-59 Personal Ringer
This feature provides another ringer pattern for the Call Terminating to the personal
number and allows the Station user to distinguish the Called Number upon hearing
ringing.
Ringing patterns are as follows:
• Ringing Pattern A: used for an Incoming Call from Central Office to the individual
number.
• Ringer pattern B: used for an Incoming Call from a Station or TIE Line Trunk to
the pilot number.
• Ringer pattern C: used for an Incoming Call from a Station or TIE Line Trunk to
the personal number.
Ringer pattern A, B and C correspond to ringer pattern 0, 1 and 5 in System Data
respectively.
P-75 Peer to Peer Bandwidth Control
This feature controls the line capacity (communication speed) to be used for particular
packets to prevent network congestion and decline in voice quality. Also, this feature
allows the System to manage the bandwidth value in use statistically. When a call is
originated or answered, the system automatically determines whether sufficient
bandwidth can be reserved or not.
Note: Location is a group of equipment accommodated to the SV7000, which is divided
by network address unit. This is a basic unit for performing LAN management (ToS
control, Bandwidth control, VLAN). Each location requires a unique ID, which is called
Location ID (LOC-ID).
P-77 Phantom Number Distinctive Ringing - Single Line (Series 8800 software
feature)
This feature (available since Release 12 software) allows Incoming Calls to a Phantom
Number to ring with a Distinctive Ringing Pattern, allowing a user to determine if the call
is for them by the Ringing Pattern.
Q-9 Quick Transfer VMM (Series 8800 software feature)
This feature permits a Dterm or Attendant Console to quickly transfers a call to another
Station’s Call Forwarding - Don’t Answer destination.
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R-2 Remote Access to System
This feature allows a user Outside the System to access the system via an exchange
Network Connection without Attendant or Station assistance. The Outside user may
originate calls over any or all of the SV7000 facilities, such as WATS, FX, TIE Line or
CCSA.
When a wrong Code is received from a Station/Trunk, detailed information on the
unauthorized user is output as
R-3 Reserve Power
This feature provides a system with a backup power supply, functioning from a battery
source, in the event of a commercial power failure.
R-4 Restriction from Outgoing Calls
This feature automatically denies preselected Station Lines within the System the ability
to place Outgoing Calls and/or certain miscellaneous Trunk Calls without Attendant
assistance.
R-5 Rotary Dial Calling
This feature permits Station users to originate calls over rotary or pushbutton lines using
Rotary Dial telephones.
R-6 Route Advance
This feature automatically routes Outgoing Calls over alternate facilities when the firstchoice Trunk Group is busy. User selects the first-choice route by dialing the
corresponding Access Code, and the equipment then advances through Alternate Trunk
Groups only if the first choice is busy.
R-7 Remote Maintenance
This feature allows Office Data changes or reassignments to be performed without a site
visit by service personnel, and can be used to detect fault tendencies before they affect
service. One REMOTE MAINTENANCE center can service an unlimited amount of
systems, thus reducing the amount of personnel needed to maintain each site.
R-16 Radio Paging
This feature allows Station users dialing access to customer-owned RADIO PAGING
equipment and to selectively tone or voice/tone alert individuals carrying pocket RADIO
PAGING receivers. The paged party may be connected to the paging party by going to
the nearest telephone and dialing a unique Answer-back Code.
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R-27 Route Restriction - Announcement
This feature permits a Station-originated Call, dialed to a restricted outgoing number, to
be automatically routed to a recorded announcement informing the caller that the Dialed
Number is restricted. (TOLL RESTRICTION [T-7])
R-37 RS-464A PAD
This service is for changing the current PAD control system for the lines and external
Trunks to a PAD control system which conforms to the RS-464A specification.
R-49 Ringer Mute
This feature allows a Station user to stop ringing by pressing the associated Feature Key
on the Dterm.
R-51 Route Name Display
This feature allows a user to assign a desired Trunk type for a route by the System Data.
The assigned Trunk type (maximum of 4 or 8 digits) is displayed on the upper line of
Dterm LCD.
•
•
•
Four-digit display
Trunk Number
Eight-digit display
R-54 Remote Call Forwarding Control
This feature allows a Station user to set/cancel the following forwarding features from
the target Station by dialing the identification (ID) Code preset for the forwarding Station.
• CALL FORWARDING - ALL CALLS [C-5]
• CALL FORWARDING - BUSY LINE [C-2]
• CALL FORWARDING - DON’T ANSWER [C-3]
• SPLIT CALL FORWARDING [S-99]
These forwarding features can also be set/canceled from an Outside Station via Central
Office/TIE Line.
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R-55 Remote Call Hold (Series 8400 software feature)
This feature (available since Release 10 software) provides the ability for a Dterm or the
Attendant Console to place a call on hold (Remote Hold), at another Dterm Station or
Virtual Extension. The original call that is placed on Remote Hold can be from an
Internal Station or Trunk.
R-57D Ringing Line Pickup - Dterm
This feature allows Dterm users to answer Incoming Calls by going Off-Hook or pressing
Answer Key. Line Preference data is to be set through the MAT (AKYD).
S-1 Single-Digit Station Calling
This feature allows the assignment of single digits for Station Numbers.
S-2 Special Dial Tone
This feature provides a distinctive Dial Tone to a Station user after the switch hook has
been pressed, enabling the user to activate a specific feature.
S-3 Speed Calling - System
This feature allows a user or ATTENDANT CONSOLE [A-3] to call frequently Dialed
Numbers using fewer digits (Abbreviated Call Codes) than would normally be required.
S-3D Speed Calling - System - Dterm
This feature allows a Dterm user to call frequently Dialed Numbers using fewer digits
(Abbreviated Call Codes) than would normally be required.
S-4 Splitting
This feature allows the ATTENDANT CONSOLE [A-3] to speak privately with one party
on an Attendant-handled Connection without the other party overhearing.
S-4D Splitting - Dterm
This feature allows a Dterm user to alternately converse between two separate parties
while one party is connected and one party remains on hold. The Station user uses the
TRANSFER button to alternate conversations between the two parties.
S-6 Station-Controlled Conference
This feature allows any NEAX2400 IPX Station to establish a Conference. The
Conference may be any combination of Stations and/or Trunks (Inside and Outside
Parties).
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S-7 Station Hunting - Circular
This feature permits a call to be processed automatically through a Hunt Group of busy
extensions, in a programmed order.
S-8 Station Hunting - Secretarial
This feature allows calls to a Hunt Group to forward to a secretarial Station when all the
Stations in the group are busy.
S-9 Station Hunting - Terminal
This feature enables calls placed to busy Stations, other than Pilot Stations of a Hunt
Group, to receive a Busy Tone rather than proceed through the normal hunting process.
However, if the call is placed to the busy Pilot Station of a Hunt Group, Station hunting
will proceed as usual.
S-10 Station Message Detail Recording (SMDR)
This feature provides a call record for all outgoing Station to Trunk Calls and Incoming
Trunk to Station Calls. When the system is equipped with this feature, a SOCKET
interface (LAN) is provided, permitting interface with a customer-owned computer
system. All output is in the ASCII format, and includes the following:
• Calling Telephone Number
• Dialed Number (maximum of 24 digits)
• Route Number
• Start of call time
• Disconnect time
• Year, month and date
• Condition (Attendant handled, transfer, etc.)
• ACCOUNT CODE [A-18] (10 digits maximum)
• AUTHORIZATION CODE [A-28] (10 digits maximum)
• FORCED ACCOUNT CODE [F-7]
[Additional FCCS Format]
• Called Party Type (Attendant Console/Station)
• Call Start/Call end time (Milli-second)
• Call metering
• FPC/User Group/Telephone Number (Calling Party)
• FPC/User Group/Telephone Number (Called Party)
• FPC of the Node providing the route for the call
• Logical Route Number
• FPC of the Node providing the first-choice route
• First-choice Logical Route Number
[Station-to-Station Basis SMDR] SMDR can be output for Station-to-Station Calls by
assigning data.
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[SMDR by MA-ID via IP Network] Multi Area ID (MA-ID) can be assigned on a network
basis for identifying the location. Then, SMDR can be output for calls over IP network
when MA-ID of both parties are not identical.
S-11 Station-to-Station Calling
This feature permits any Station user to directly dial another Station within the system
without operator assistance.
S-12 Station-to-Station Calling - Operator Assistance
This feature allows a Station user to call another Station within the System, with the
assistance of the ATTENDANT CONSOLE [A-3] operator.
S-13 Step Call
This feature allows the Attendant or Station user, upon calling a busy Station, to call an
Idle Station by dialing an additional digit. This feature will operate only if the number of
the Idle Station is identical to that of the busy Station
in all respects, except the last digit.
S-15 Serial Call
This feature allows the Attendant to arrange for a recall from a Station before releasing a
Central Office Call to that Station. When the Station subsequently disconnects from the
call, the Central Office party automatically rings back to the Attendant.
S-19 Single-Digit Feature Code
This feature allows the system to be programmed so that features used most frequently
may be accessed by dialing a single digit.
S-21 Speed Calling - Station
This feature allows a Station user to dial certain frequently-called exchange network
numbers using fewer digits (Abbreviated Call Codes) than normally required. Using this
feature, a Station user can establish his own abbreviated codes.
S-23 Speed Calling - Group
This feature allows a Station user to share a set of common Speed Calling Numbers with
other Station users in the group.
S-24D Save And Repeat - Dterm
This feature allows for a Dterm set to save a specific Dialed Number and then redial that
number later.
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S-25D Service Display - Dterm
This feature generates LCD displays corresponding to the various features as they are
initiated.
S-26D Speed Calling - One Touch - Dterm
This feature allows Dterm users to dial Telephone Numbers by pressing a Single Key.
Up to 12/24 numbers may be programmed by the Dterm Station user.
S-27 Service Feature Peg Count
This feature provides a statistical usage report on various features, generated at the
MAINTENANCE ADMINISTRATION TERMINAL (MAT) [M-18]. This report will provide
the Communications Manager with detailed users’ habits and planning data for the
communications system.
S-33 Software Line Appearance
This feature permits ports that do not physically exist to be used as Sub Lines on Dterm
Stations. Ports normally reserved on a Dterm for data Stations only, can be assigned as
a SOFTWARE LINE APPEARANCE. Additionally, 64 software ports have been added to
a Port Interface Module. The 64 software-assigned ports can be assigned as Virtual Line
Appearances on Dterm Stations. The use of the data ports and the 64 software-assigned
ports conserve valuable hardware for Stations and Trunks.
S-61 Speed Calling Override - System
This feature allows numbers programmed for SPEED CALLING to be available to
Stations on a system-wide basis, determined by Service Feature Class (SFC). For
example, SPEED CALLING-SYSTEM [S-3] numbers, available to Stations that are not
restricted, can be made available to Stations that are restricted.
S-62 Station 5db PAD
This feature allows a Single Line Station user to be affected by the 5db loss through the
system on Station-to-Station Calls.
S-64 Serial Call - Loop Release
This feature allows an ATTENDANT CONSOLE [A-3] Loop Key to become available
after setting SERIAL CALL [S-15].
S-99 Split Call Forwarding
This feature allows a Station to set two different target Stations for CALL
FORWARDING-ALL CALLS (FORWARD)/BUSY LINE (FORWARD-BY)/DON’T
ANSWER (FORWARD-DA), depending on whether the Incoming Call is from an internal
Station or an Outside Party (Central Office TIE, DID, etc.).
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S-106 Slumber Time - Do Not Disturb
This service allows a slumber time for up to four times per day on a Station Group basis.
During the slumber time, all Stations in the group concerned are placed into “Do Not
Disturb” mode; and Incoming Calls (Station/DID/DIT) are routed to the Attendant or an
Announcement Machine.
S-127 Serial Call - Dterm
This feature allows the Dterm user to set SERIAL CALL to the central office line/TIE Line
Call and extend the call to the desired Station/Trunk. When the communication with the
desired party ends, the announcement instructing the caller to dial the next desired party
number is sent out. The caller, by following the instructions, can be connected to a
number of destinations without hanging up.
S-129 SMDR Output Expansion - ANI/CPN
This feature expands SMDR information output, by which Calling Numbers from Central
Office may be output to the expanded area.
S-137 Selectable Station Call Pickup
This feature enables the Station/PS user to pick up the Incoming Call to Station, which
are not specified by Service Feature Restriction Class (SFC) data, within Call Pickup
Group/Call Pickup Expand Group by dialing Selectable Station Call Pickup/Selectable
Station Call Pickup Group Access Code. Incoming Call to pre-specified Stations/PSs
cannot be picked up.
S-138 Station Hunting - Priority
This feature permits a call directed to a Busy Station of a Hunting Group to be
automatically forwarded to other Stations within the Group, in a programmed order. At
this time, the call is forwarded from Higher Priority Stations to Lower Priority Stations.
When the Lowest Priority station is also busy, the caller receives Busy Tone.
S-139 Station Hunting - Priority with Switchback
This feature permits a call directed to a Busy Station of a Hunting Group to be
automatically forwarded to other Stations within the Group, in a programmed order. As a
first step, the call is forwarded from Higher Priority Stations to Lower Priority Stations.
S-140D Side Tone On/Off - Dterm
This feature allows the Station (both IP Station and Non-IP Station) to switch Side Tone
ON and OFF.
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S-141 Survivable Remote MGC
Survivable Remote MGC (SR-MGC) is placed on remote office to manage the call of IP
terminals and trunk call to/from PSTN via MG in case of network failure or SV7000
breakdown on the main office. In normal state, SR-MGC operates in stand-by mode
under monitoring of the main office SV7000. Once the SV7000 cannot accept the
registration of IP terminals in remote office, the operation is switched over to the SRMGC. When the main office SV7000 recovers, the SR-MGC returns to stand-by mode
again.
S-143 Software Multiple My Line Appearance
This feature permits a Dterm/Dterm IP User to assign the same My Line numbers to the
Sub Lines, which will be able to originate and terminate a call by the Sub Lines while
another call is being originated or terminated by the My Line. A Sub Line assigned to the
same My Line number is mentioned as the Same Number Sub Line afterwards.
S-144 SMDR Multi Area ID via IP
This feature allows the system to record the billing information that terminal user makes
a call via Media Gateway (MG).
T-1 Tandem Switching of TIE Trunks - 2/4-Wire
This feature allows Trunk-to-Trunk Connections through the switching system, without
the need for any Attendant assistance or control. The major use of this feature is in
association with the Dial Tandem TIE Line Network to allow TIE Line Connections and
Incoming TIE Line Calls automatic access to, and completion of, local Central Office
Calls.
T-2 Three-Way Calling
This feature enables any Station user to add another party to an existing Connection,
establishing a Three-way Conference. Even if the additional Dterm Station user holds
the call, THREE-WAY CALLING may be established. (This service is called Consultation
Hold.)
T-2D Three-Way Calling - Dterm
This feature enables a Dterm user to establish a Three-way Conference by connecting
an additional party to an already existing conversation. Even if the additional Dterm user
holds the call (Consultation Hold), THREE-WAY CALLING may be established.
T-3 TIE Line Access
This feature allows any Station user dial access to a TIE Line.
T-5 TIE Line Connection With Pad Control
This feature provides a switchable transmission pad on TIE Trunks that allows Tandem
Connections. Necessary pad control is activated to protect against echo.
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T-6 Toll Denial/Toll Diversion
This feature prohibits Station users from placing Long Distance Calls over a specific
group of Trunks. Restricted calls are routed to either an ATTENDANT CONSOLE [A-3]
(diversion) or to an intercept tone (denial).
T-7 Toll Restriction - 3/6-Digit
This feature allows the System to be programmed to restrict Outgoing Calls according to
specific area and/or central office codes. This restriction is controlled on the basis of a
three-digit Area Code or a six-digit combination area and Office Code Numbering Plan.
T-8 Trunk Answer from Any Station (TAS)
This feature allows any Station (except one with incoming restrictions) to answer
Incoming Calls when the system is in Night Mode. Incoming Exchange Network Calls will
activate a common alert (TAS) signal at the customer’s premises. By dialing a specified
Code, any Station may answer the call and then extend it to any other Station using
CALL TRANSFER-ALL CALLS [C-11].
T-9 Trunk Group Busy Lamp
This feature provides the ATTENDANT CONSOLE [A-3] with a visual indication when all
Trunks in a particular Trunk Group are busy (LDN, WATS, FX, DOD, CCSA, TIE or
Special Trunks). Assignment of the TGBLs is made using the MAINTENANCE
ADMINISTRATION TERMINAL (MAT) [M-18].
T-10 Trunk-to-Trunk Connection
This feature provides any Station user with the ability to Conference together either two
Central Office Calls or a Central Office and TIE Line Call.
T-12 Tenant Service
This feature provides for more than one organization (Tenant) to share the same System.
Through system programming, each organization may be restricted to its own Central
Office Trunks, ATTENDANT CONSOLEs [A-3], and extension links. In addition,
Incoming Calls are directed to the specific organization (Tenant).
T-18 Time Display
This feature provides a digital time display on the Attendant Console.
T-18D Time Display - Dterm
This feature provides a digital time display on a Dterm’s LCD.
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T-23 Trunk Line Appearance
This feature allows a Dterm user to directly access a Central Office Trunk or TIE Line
without dialing an Access Code. Incoming Calls can also be answered at the TRUNK
LINE APPEARANCE [T-23].
T-28 Trunk-to-Trunk Third-Party Cancellation
This feature allows a Station to cancel an active Trunk-to-Trunk Connection (during a
Three-way-Connection) and return to the Original Call.
T-62 Timed Reminder
This feature allows a Station user to set a Timed Reminder Call from the Station. The
system calls up the Station at the designated time. When the Station answers the call,
the Station user hears a message or Music-On-Hold.
T-63 Tone Block
This feature allows the user of a Station (via analog adapter) or an Analog Station (PB)
to restrict any call. This prevents interruption of data transmission, before the Connection.
U-1 Uniform Call Distribution (UCD)
This feature distributes Incoming Calls to a UCD Group of up to 100 Stations. Calls are
distributed to Idle Stations in a circular pattern, in the order in which they arrive.
U-3 Universal Sender
This feature permits Dialed Numbers to be transmitted via a Sender, a device that will
automatically add or delete the necessary number of digits or perform any necessary
conversions.
V-1 Variable Timing Parameters
The Variable Timing Parameters feature gives the System the versatility to change
timing parameters, using the MAINTENANCE ADMINISTRATION TERMINAL (MAT) [M18]. All timing parameters for a specific timing sequence are set initially within the
generic program. These timing parameters can be changed to accommodate user
requirements.
V-2 Voice Call
This feature enables a Single Line Telephone to make a VOICE CALL [V-2] when the
Called Party is a Dterm set. This path exists from the Calling Party to the Called Party's
built-in speaker. If the Called Party's MIC is ON, the Called Party can converse handsfree.
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V-2D Voice Call - Dterm
This feature provides a voice path between two Dterm sets. This path exits from the
Calling Party to the Called Party’s built-in speaker. If the Called Party’s MIC Key is ON,
the Called Party can converse hands-free.
V-3D Volume Control - Dterm
This feature is used for controlling the volume of a Dterm set’s built-in speaker and
receiver.
V-25 Voice Over IP H.323 Connectivity
This feature allows the system to use the voice communication adhered to H.323
standard protocol between the SV7000 and a terminal adhered to the H.323 protocol. To
use this feature, the System requires the IPTRK Circuit Card.
V-26 VS32 Conference Server
The VS-32 is a voice server with Announcement feature, External Hold Tone feature and
Conference Feature. These features can be used over the networks. Communication
cost can be reduced drastically because all of these features are operated on the IP
network. The VS-32 Conference Server can operate these features on SV7000 without
having any Circuit Card or PIR. For more detailed information, refer to the IP Peripheral
Equipment Guide.
W-1 WATS Access
This feature permits any Station user direct-dial access to outgoing Wide Area
Telephone Service (WATS) lines.
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