VoiceCon Spring 2007 - Conference Presentation

VoiceCon Spring 2007 - Conference Presentation
Avaya Response to:
VoiceCon 2007
Request for Proposal for:
IP Telephony System
Presented by:
Jack Hilbert & Carlo De Luca
Global Response Manager(s)
(720) 444-7910
[email protected]
(908) 953-5755
[email protected]
December 1, 2006
06
©2006 Avaya Inc.
Page 1
VoiceCon
Request for Proposal for an IP Telephony System
Preface
Preface
The following RFP document was exclusively designed and developed by
TEQConsult for the VoiceCon® Spring 2007 Conference.
The RFP is intended to solicit product information and pricing data about IP Telephony
systems during the Fall 2006 time period. The RFP was written for a large multi-facility
enterprise configuration with IP voice terminals as the primary station user interface to
the system. TEQConsult Group recognizes that every business and institution has unique
communications needs and resources, but the much of the material included herein can
be used by VoiceCon workshop attendees regardless of their unique system size and
application requirements.
VoiceCon workshop attendees may use this RFP as a template for customizing their own
RFP with the proviso that proper accreditation to TEQConsult Group will be included in
the document.
TEQConsult Group would like to thank Fred Knight, VoiceCon GM and the publisher of
Business Communications Review, for his review and editing of this document and to
Unimax Systems Corporation for its contributions to the systems management section of
the RFP.
Avaya Response:
Read and understood.
VoiceCon IP Telephony System Request For Proposal
General Guidelines for Proposals
1.
Please read though the entire RFP before beginning to work on your response.
2.
Configure and price your system design to satisfy all stated RFP requirements,
including any and all system hardware and software elements necessary to
satisfy a requirement. Vendors that underconfigure their system design to
reduce its price proposal will be penalized.
3.
All products and solutions proposed for this RFP must be formally announced as
of January 15, 2007 (prior to VoiceCon Spring 2007).
4.
Do NOT provide material or information unrelated or not relevant to a specific
RFP clause requirement.
5.
Be brief, but complete, in your responses.
6.
Provide succinct, clear, and unambiguous responses; do not obfuscate your
responses with unnecessary wordage.
7.
Make sure to review and edit your proposal before submission.
8.
All proposals are due December 1, 2006. Late submissions and/or revisions to
submitted proposals may not be accepted. Deadline extensions may be granted
under acceptable circumstances, only.
Avaya Response:
Read and understood.
06
©2006 Avaya Inc.
Page 1
VoiceCon
Request for Proposal for an IP Telephony System
Preface
Proposal Evaluation
The proposals to the RFP will be judged on the following factors:
1.
Satisfaction of system performance requirements
2.
Price of the proposed solution
3.
Adherence to each of the above general proposal guidelines
Avaya Response:
Read and understood.
Important submission requirements:
Submit Part 1 System Performance Requirements responses in MS Office
WORD file format, excluding responses to RFP Clauses specifying
PowerPoint format, e.g., Clause 1.0.1. When PowerPoint format is
requested do not copy/paste PDF format graphics or images.
Submit Part 2 System Pricing responses in MS EXCEL file format
Avaya Response:
Read and understood.
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
PART 1:
System Performance Requirements
Submit Part 1 responses in MS Office WORD file format except when otherwise
noted.
Avaya Response:
Comply.
1.0.0 System Overview
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 20% 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).
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
Figure 1
Voicecon IPTS Network
HQ IPTS
1200 stations
SB2 SRG
50 Stations
SB1 SRG
100 Stations
WAN
RO IPTS
SB3 SRG
10 stations
December 1, 2006
HQ: Headquarters
RO: Regional Office
SB: Satellite Branch
IPTS: IP Telephony System
SRG: Survivable Remote Gateway
©2006 Avaya Inc.
250 Stations
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
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
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
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
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
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 1Gbps 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) RJ-45 wall connectors to the local
wiring closet. The RJ-11 and RJ-45 connectors will be either wall mounted or mounted
in the modular furniture throughout the office environment. 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.
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
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 MS PowerPoint format (identify the
file as part of your electronic proposal submission), and also copy/paste here the
diagram in the submitted MS WORD file proposal.
Proposed IPTS Network Diagram Here
Avaya Response:
Comply; please also see Avaya Appendix 3- PPT Slides for additional drawings.
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
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.
Avaya Response:
The proposed Avaya Communication environment is open and standards based –
essentially network infrastructure agnostic, for real-world flexibility. In the complex
world of IP Telephony, we understand that enterprises often “inherit” LAN/WAN
infrastructures from multiple vendors via mergers and acquisitions. Avaya has
talented engineers in-house who are qualified to assess, evaluate, and recommend
solutions that will protect your current infrastructure investment as well as support
future requirements. We recommend that every client planning to install an IP
Telephony solution have a Network Assessment and Recommendation engagement
with our Avaya Global Services Organization.
Avaya has done successful interoperability testing with several major LAN/WAN
infrastructure vendors, including Cisco, Nortel Networks, Juniper, and Extreme
Networks. More information on this and additional system testing is available upon
request.
1.1.0 Basic Communications System Requirements
The proposed IPTS should be in current production and operating as a commercial
system for at least five (5) customers in the USA.
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 2006 to be accepted by VoiceCon in response to this RFP.
This is a mandatory requirement to submit a RFP response.
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
Avaya Response:
Comply with clarification. The proposed solution is in production and operating for
numerous customers throughout the USA and Internationally. We have provided
corporation names and the other requested information here, however, individual
contact names and telephone numbers have been provided to Allan Sulkin in a
separate PROPRIETARY – RESTRICTED document, due to the web posting and mass
distribution of this proposal. Names and telephone numbers of individual contacts
are available on an as needed basis. Please contact your Avaya Client Executive or
send an email to [email protected] for additional information.
Reference Companies:
Customer Name
Date of
Installation
Apollo
Group/University of
Phoenix
2005
Charter Steel
2006
Australian National
University
2003
QualChoice
2002 through
2003
Winterthur
2005
December 1, 2006
Current System Release
Avaya Communication Manager,
Release 2.0
Avaya Modular Messaging
Avaya Interaction Center
5,000 Avaya IP Telephones with a total
of 20,000 endpoints across the United
States.
Avaya Communication Manager,
Release 3.0 over 5 locations
Avaya Communication Manager,
Release 1.3
15,000 IP telephones over 130 building
campus
Avaya Communication Manager,
Release 2.0
Contact Center
900 IP telephones over 3 locations
Avaya Communication Manager,
Release 2.2
Avaya Modular Messaging
2,200 Avaya IP telephones over 22
locations
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
1.1.1 Single System Image
The proposed system must provide a true Single System Image across VoiceCon HQ
and remote facilities locations to include:
1) 5-digit dialing between all stations;
2) 100% transparent operation across VoiceCon facilities for all station, attendant, and
system 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
support of 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 answers to each of the following questions:
1. Is the proposed IPTS solution a single system solution or a networked multiple system
solution?
2. Does the proposed IPTS solution fully satisfy all six (6) of the stated Single System
Image requirements? If not, explain which of the requirements are not satisfied?
Avaya Response:
Comply. The proposed solution is a single system, controlled by a centralized S8720
Media Server stack at the Headquarters location. The solution supports the six
Single System Image requirements as follows:
1. 5-digit dialing – since one centralized server controls the design, all extensions
will be assigned and controlled by the S8720 Media Server at the HQ location.
The Regional Office (RO) will contain an S8500 Media Server running in
Enterprise Survivable Server (ESS) mode; this Media Server will be able to run
the whole Enterprise (total licences purchased at the HQ location) in the event of
a catastrophic failure at the HQ location.
Each of the Branches (SB1, SB2) will have an S8300 Local Survivable Processor
(LSP) with the smallest branch (SB3) running a G250, they can run independent
during a network failure or a HQ/RO catastrophic failure, they will obtain
translations (programming) from the HQ S8720 Media Server. The two larger
S8300 LSP branch locations can support up to 450 IP endpoints each if required
during a prolonged outage
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
A major advantage of the proposed solution is the ability of the locations to have
the same extension at multiple sites. For example, if associates travel between
locations, it would be convenient for the Mail Room and/or Guard’s desk to have
the same extension at all locations. The Multi-location dialing feature of Avaya
Communication Manager supports this functionality. In addition, if the branch
locations are currently autonomous, there may be duplicated extension numbers
at several locations. Multi-location dialing minimizes user disruption by
minimizing the extension numbers that have to be changed.
2. A High Level of Transparent Operation for Commonly Used Station, Attendant,
and System Features – since the solution is a single system, transparent
operation and features is an inherent capability.
3. Centralized Systems Management and Maintenance Operations Using a Single
Unified Customer Database for All Location Equipment and Station Users – the
proposed solution includes Avaya Network Management. This group of
applications allow users to access and manage the system from anywhere on
the network, or a remote location. Users will have access to any elements of the
entire network, assuming they have the appropriate access credentials.
4. Avaya recommended attendant services supports the required capabilities for a
centralized attendant operation (pooled or non-pooled) that supports all
locations without loss of feature functionality; including station busy indication,
call diversion, call status, emergency access, call display, call transfer and more.
5. Shared Messaging System Resources – the proposed solution includes Avaya
Modular Messaging for all stations on the network. All users will have exactly the
same features, capabilities, and access.
6. Automatic Alternative Routing Across the Network for IP and Circuit Switched
Trunk Calls – Avaya Communication Manager supports many routing options,
which will all be considered and implemented according to VoiceCon’s
requirements at installation.
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
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.
Vendor Response Requirement:
Confirm that the proposed communications system solution supports E911 service for all
user stations (IP and analog) at 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?
Avaya Response:
Avaya has a history of leadership with regard to public safety and currently
supports E-911 for traditional “fixed” endpoints and IP hard-phones and IP softphones. Working with RedSky Technologies, Avaya provides a complete solution for
desk-top level identification for E-911 supporting digital, analog and IP phones.
As a basic requirement for E-911, the Communication Server must be able to outpulse a unique 10-digit DID number over the PSTN utilizing either ISDN-PRI or
CAMA trunks. Avaya systems comply with this requirement. The other basic
requirement is to identify the location of the caller by populating the regional ALI
database that serves your geographic area. The level of granularity for location
identification is based on the methodology employed by each PBX owner and can
range from desktop identification to network region identification.
In order to ensure proper location identification, it is critical that the ALI database
be maintained with up to date information. RedSky’s E-911 Manager is tightly
integrated to the Avaya system to automatically capture location changes and
update the ALI database on an ongoing basis. The requirements for ALI records
vary across the United States and are based on the Local Exchange Carrier that
serves each region. RedSky’s E-911 Manager is pre-configured to submit ALI
records based on these disparate requirements, thus seamlessly supports
nationwide enterprises from a single server.
December 1, 2006
©2006 Avaya Inc.
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Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
How E-911 works for traditional end-points:
For traditional end-points, the telecom administrator updates location information
for every move, add or change on page 3 of the station screen. RedSky’s E-911
Manager interfaces to the Avaya S8720 Media Server to capture the location
information from the building, floor and room fields for each desktop, and
subsequently updates the ALI database with valid records in the required format.
How E-911 works for IP phones:
IP phones can be managed in two ways depending on your network configuration –
either by network region/subnet or by port.
Identification by Network Region:
As part of the initial configuration of the S8720 Media Server, the customer will be
required to define network regions/subnets geographically and assign ranges of IP
addresses to each region. Each IP address range will have an assigned Emergency
Location Identification Number (ELIN). The ELIN will be the 10-digit DID number to
route the 911 call. Each ELIN must have an associated ALI record at the regional
ALI database that represents the network region or Emergency Response Location
(ERL) of the caller.
The PBX owner is responsible for establishing and maintaining updated and
accurate records in the ALI database for each DID number or ELIN. RedSky’s E-911
Manager automatically interfaces to the S8720 Media Server to capture the location
records, translate them into valid NENA records and update the regional ALI
database on an ongoing basis. By automating the process with RedSky’s software,
administrators eliminate any human error in updating the regional ALI database
manually every time there is a change.
E-911 Location for IP Phones Defined by Network Region
1
Floor 20
DHCP Server
Region
Region 1
2
IP Phone
Floor 19
Region 2
Floor 18
Region 3
DHCP Config
IP Range
Region 1
Region 2
Region 3
192.168.1.1-254
192.168.2.1-126
192.168.2.127-254
RedSky E-911
Manager App
Server
IP Phone
Routers w/ DHCP Overlay
3
Communication
Server
Region
1
2
3
IP Address Mapping
IP Range
192.168.1.1-254
192.168.2.1-126
192.168.2.127-254
ELIN
312-555-5555
312-666-6666
312-777-7777
5
Modem
Connection
RBOC
Gateway
ISDN Prime or
CAMA Trunk
through Local
C/O
4
RBOC ALI
Database
PSAP
December 1, 2006
©2006 Avaya Inc.
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Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
1. Network Regions are defined geographically (Region 1=Floor 20, Region 2=Floor
19, etc.) Each region is serviced by a router or a router port that has DHCP
relay capability.
2. Corresponding IP address ranges are established for each region in the DHCP
server and in the IP Network Mapping Form in the Communication Manager 3.1.
Therefore, when an IP phone is plugged in, it is issued an IP address for that
Region by the DHCP server.
3. RedSky’s E-911 Manager creates an ALI record for each Region (Bldg, Address,
Floor, and Room) and associates it to the ELIN contained in the IP Network
Mapping form.
4. E-911 manager forwards the ALI records to the regional ALI databases which
forward them to the PSAP.
5. When a 911 call is made from any phone in a region, the ELIN for that region is
routed over 911 trunks to the PSAP. The ELIN prompts data retrieval for the
corresponding ALI record. Emergency crews are dispatched.
December 1, 2006
©2006 Avaya Inc.
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Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
Identification by Port:
For customers that don’t utilize the network regions in Communication Manager 3.1
(CM), RedSky’s E-911 Manager utilizes new network discovery components to
identify IP phones down to their port and switch on the network. A network matrix
is established in E-911 Manager to associate the port, its physical location, and the
network device with the ELIN. Each port will have an Emergency Response
Location, its physical location, which becomes the ALI record in the regional ALI
database. E-911 Manager communicates with Avaya Communication Manager
through a real-time interface to detect IP phone registrations on the network. E911 Manager captures the MAC address, port and network device of the user and
references its internal network matrix to define the appropriate ELIN for 911 calls.
E911 Manager overwrites the new ELIN to the ELE field of the station in the Avaya
call server. This ELIN will be out-pulsed for 911 calls for this user and will have a
corresponding record in the regional ALI database indicating the caller’s location.
The ALI database is updated on a regularly scheduled basis to ensure database
synchronization with the call server. This is very important to capture any changes
on the network. As an additional safety measure, if a port has not yet been defined
with a location, the E911 Manager will identify the gap, designate the next level of
granularity for E-911 identification and alert the administrator.
E911 Location for IP Phones Defined By Port
December 1, 2006
©2006 Avaya Inc.
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Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
A network matrix is established in E-911 Manager with defined Emergency
Response Locations (ERL) that are associated with Emergency Location
Identification Numbers (ELIN). Ports and switches are assigned to each defined
ERL.
1. ALI records are created in E-911 Manager for each ERL/ELIN association and
updated in the regional ALI database through E-911 Manager’s automated
interface.
2. E-911 Manager is automatically notified via a real-time interface with the
Avaya S8720 Media Server every time a phone registers on the network.
3. Using the MAC address, E-911 Manager captures the MAC address and the
network device port and establishes the appropriate ELIN in the Avaya
Communication Server in anticipation of a 911 call.
4. When a phone dials 911, Communication Manager will out-pulse the assigned
ELIN to the Public Safety Answering Point (PSAP). The ELIN prompts a data
retrieval for the corresponding ALI record. Emergency crews are dispatched.
For nomadic IP Softphone users operating out of jurisdiction of the main call server,
RedSky offers Location Information Services (LIS) as an optional feature on E911
Manager. LIS enables national E911 calling and location identification for
enterprises supporting nomadic IP Softphone users. The existing 911 network
cannot support dynamic call routing for users that are outside the 911 jurisdiction
of the central call server. In working with 911 industry partners, RedSky now offers
Location Information Services that capture the location of remote IP phone users as
they log on to their Softphone.
E911 Manager with LIS captures the caller’s location as the Avaya IP Softphone
registers on the network. E911 Manager updates the VoIP Positioning Center (VPC)
database with the location of the remote user so that 911 calls can be properly
routed to local Public Safety Answering Point (PSAP) serving the caller.
LIS options for capturing location information:
Network Discovery -- On-premise gateway or discovery device captures phone
registrations on corporate sites and enables emergency location updates.
Softphone Location Determination Application – A RedSky client application
that runs on the same device as the Avaya IP Softphone client that prompts users
for location information before access to phone service is granted.
December 1, 2006
©2006 Avaya Inc.
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Part 1 - System Performance Requirements
LIS Web Application – A Web-based application accessed by end-user to
establish location information.
LIS captures the location data from the user, validates it against the Master Street
Address Guide (MSAG) and submits it to the VPC in anticipation of a 911 call. If 911
is dialed, the call server will out-pulse the 10-digit number of the dialing phone
directly to the VPC. The VPC will route the call to the local PSAP serving the caller
based on the location information it received from RedSky’s LIS. This feature is
targeted to serve nomadic IP Softphone users and branch offices that do not have
local 911 trunks. LIS is available as a monthly service based on the number of
designated users. See Appendix B, System Requirements for more information.
¾ Supports local 911 calling anywhere in the USA for Softphone users and
branch offices
¾ Eliminates the need and cost of local 911 trunks for branch offices. Trunk
cost savings can be significant for situations with hundreds of branches
¾ Fulfills corporate responsibility for providing E911 services to all phone users
by requiring the Softphone user to identify their location every time.
This diagram shows how nomadic IP Softphone users can designate their
location and receive complete E911 calling coverage.
Why Avaya and RedSky:
Avaya and RedSky deliver comprehensive and technically advanced E-911 solutions
for VoIP networks. Customers like VoiceCon can select from numerous E-911
options including network regions, network discovery, or station screen location ID
depending on their building and network configurations. Avaya IP soft phone
technology allows the user to supply a “real” landline location so that emergency
calls are routed to the correct emergency responder and a valid location is
provided. The highly reliable Avaya systems provide confidence that 9-1-1 calls will
be processed regardless of outages that may occur due to a link failure or an
outage at a remote site. Full-featured survivable remote gateways will also provide
PSTN access and/or E911 capabilities, even in the event of an IP WAN router
failure.
December 1, 2006
©2006 Avaya Inc.
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Part 1 - System Performance Requirements
RedSky’s E-911 Manager is a fully automated system designed for large enterprises
with multi-switch environments. It is completely scalable to support hundreds of
PBXs/communication servers with traditional and IP endpoints. By automating
critical tasks and integrating to the enterprise phone system, E-911 Manager is a
highly cost-effective and reliable solution.
Additional Information:
Avaya has been supporting E-911 functionality for many years and continues to
enhance the operation by leveraging the strengths of our development partners.
With these partnerships, Avaya provides reliable, cost effective and comprehensive
E-911 solutions. With Avaya systems, customers can migrate to IP telephony while
continuing to leverage existing infrastructure, offering a migratory transition.
In addition, the Avaya solutions are built on open standards and are network
infrastructure agnostic – they do not depend on proprietary IP network hardware.
Avaya also offers other capabilities to alert others of E911/911 emergency calls,
including:
¾
Alerting to consoles and stations
¾
Callout to digital pagers
¾
Automatic call recording
Leveraging the Avaya CMAPI interface, RedSky’s Emergency On-site Notification
(EON) feature notifies on-site personnel with the exact location, extension and
name of the 911 caller. This feature is essential to improve emergency response for
those facilities with on-site security.
December 1, 2006
©2006 Avaya Inc.
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Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
E-911 Manager V5.0 System Requirements
The server to support E-911 Manager, V5.0 is supplied by VoiceCon. The System
Requirements are:
System Requirements E911 Manager Server
E911 Manager Server
Hardware/Software
Recommended Minimum
Processor(s)
Pentium IV, 2.4 GHz
RAM Memory
1 GB
Hard Disk (free
space)
Network Adapter
CD-ROM
Modems
Operating System
Tape Backup
Software
Remote Access
Software
Network Services
MS Message
Queuing
Microsoft Internet
Information
Services (6.0)
ASP .NET 1.1
.NET Framework
30 GB (SCSI, RAID 5 preferred)
100 Mbit
Yes (for s/w install)
At least (1) External 56 or higher
Kbps US Robotics for transferring
data to the E911 DB Provider
*NOTE: 2nd modem recommended
if remote support will be via
modem.
MS Windows 2003 Server SP1
Yes (if tape backup used)
1. VPN client or similar access.
2. PcAnywhere v11.0 or higher, for
remote support*.
*If broadband or high speed access
is not an option, an analog modem
and pcAnywhere are required.
Remote Access Service (RAS)
V5.1 or higher (Windows Add-on
component)
IIS version 6 or higher (Windows
Add-on component)
Version 1.1 (Windows Add-on
component)
Version 1.1 (Windows Add-on
component)
Adobe Acrobat
Reader
To view reports in PDF format
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
Client Workstations
Hardware/Software
Recommended Minimum
Processor(s)
RAM Memory
Hard Disk (free space)
Network Adapter
CD-ROM
Floppy Disk
Pentium III, 500 MHz
512 MB
2 GB
100 Mbit
Yes (for s/w install)
1.44 MB
Any Windows based desktop able to run MS
Internet Explorer 6.0 or greater
Operating System
Software Notes:
E911 Manager uses ADAM LDAP as the data store. ADAM LDAP is a license free
LDAP directory from Microsoft. RedSky Technologies installs ADAM LDAP on the
server when we install the E911 application on the server.
E911 Manager Client Workstations
E911 Manager is administered via a Web Browser. Any Windows based desktop
able to run Microsoft Internet Explorer 6.0 or higher can be designated and
configured to administer E911 Manager.
E911 Manager Network Discovery Requirements
E911 Manager uses an internal capability called Real Time Reflection (RTR) to read
and write data to the switch. RTR allows data from the switch to be formatted in an
LDAP data structure and stored in E911 Manager’s LDAP data store. RTR is an
internal feature of E911 Manager and is included with the base software when
Network Discovery is purchased.
Configuration Specifications for RTR
¾ No additional servers are required to run Network Discovery
¾ RTR is a service that runs on the RedSky E911 Manager server and can
interface to up to 50 call servers that are all connected via a WAN to the
E911 Manager server.
¾ T-1 bandwidth is recommended for RTR to interface to each call server over
the WAN to optimize performance.
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
E911 Manager EON Requirements
Client Workstation requirements: (For any workstations that would like to
be notified when a 911 call is made)
Hardware/Software
Recommended
Processor(s)
RAM Memory
Hard Disk (free space)
Monitor
Network Adapter
Keyboard
Mouse
CD-ROM
Pentium 4, 2.4 GHz or higher
512 MB
1 GB (for EON client)
17” SVGA, 800 x 600
100 Mbps
101 Enhanced
Standard 3-button
Yes (for s/w install)
Operating System
Windows XP Pro or Windows 2003
Audio Card w/
Speakers for EON
audio notification
Compatible sound card
EON requirements will vary based on the type of PBX in use. See below
Requirements for Avaya S8x00 series switches with Communication
Manager 2.2 and higher:
¾ EON establishes a persistent telnet connection to Avaya Communication
Manager to monitor 911 calls.
¾ A CLAN connection is required. The CLAN can be shared with other
applications.
1.1.2.1
E911 and Station Moves
Vendor Response Requirement:
Are station user moves behind the proposed IPTS tracked dynamically in real time for
E911 services support? If not, how often is the database updated?
Avaya Response:
E911 calls are tracked and identified dynamically as described in the
“Identification by Network Region” and “Identification by Port” sections in
1.1.2 above.
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
1.2.0
Proposed Communications System Design
The proposed communications system may only be based on either of the two
following architecture technology designs:
z
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)
z 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.
Avaya Response:
Comply. Please refer to the diagram provided in Appendix 3, PowerPoint
Illustrations. The Proposed IP telephony system is the Avaya S8720 Media Server,
powered by Avaya Communication Manager, Release 3.1. The messaging system is
Avaya Modular Messaging with Speech Access, Release 3.
The proposed solution is a single system design with the centralized common
control installed at HQ location: all call processing operations for Branch/Remote
locations are dependent on the HQ location except in survivable mode. The call
processing is centralized.
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
At the Media Gateways used to support the station/trunk call processing and
switching operations are listed below:
Location
Media Gateway
HQ
Avaya G650 Media Gateway
RO
Avaya G650 Media Gateway running off the HQ S8720
Media Server with S8500 Media Server as an
Enterprise Survivable Server (ESS)
Large and Medium
Branch Offices
Each office has Avaya G700 Media Gateways with an
S8300 Media Server LSP at each office
Small Branch
Offices
Each office has a G250 Media Gateway
Avaya Communication Manager, Release 3.1,
Avaya Modular Messaging, Release 3.0
Avaya Integrated Management, Release 3.2
Dates the products were first released are listed below:
Product
Date of First Release
Avaya Communication Manager
Avaya Modular Messaging
May 6, 2002
January, 2003
Avaya S87xx Media Server Series
May 6, 2002
Avaya S8720 Media Server
February 2006
Avaya G650 Media Gateway
December 8, 2003
Avaya G700 Media Gateway
May 6, 2002
Avaya G250 Media Gateway
June 6, 2005
Avaya S8300 Media Server
May 6, 2002
Avaya Integrated Management
May 6, 2002
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
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 (serverembedded, 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
Avaya Response:
Comply. The proposed solution is a combination of the Converged and the Pure
Client Server. At the HQ site, G650 Media Gateways house port interface circuit
cards (circuit packs), communication to the server is via IP. At the Regional and
Branch locations, G700 and G250 Gateways house the signaling interface cards
(media modules) to support analog stations and trunks, and digital trunks. A brief
description of each Media Gateway type and the associated circuit packs or media
modules is provided below:
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
Avaya G650 Media Gateway
¾
AC/DC Power Supply
¾
Supports Redundant Load Sharing Power Supplies
¾
Can Support 14 Circuit Packs
¾
Up to five G650s per Port Network
¾
8U 14 high X 17.5 wide X 22 deep
REAR
FRONT
SLOT 1
655A
Power Supply
SLOT 14
Redundant 655A
Power Supply
AC Connectors
(2 Places)
Ground
DC Connector
Avaya G700 Media Gateway
The G700 Media Gateway is designed to be scalable and offer numerous options. It
is functional on its own or with other G700 Media Gateways and/or the Avaya P330
devices such as the P333T, P333R, and P334. A maximum of 250 G700 Media
Gateways can be supported using the proposed Avaya S8720 Media Server.
To provide power to IP telephones without additional cables, VoiceCon can add the
optional Avaya P333T-PWR to the same stack as the G700 Media Gateways.
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
The following list describes the basic architecture of the G700 Media Gateway:
¾
Intel i960 controller that hosts all of the base switch-control and
management software
¾
Fits in an EIA-310-D standard 19-inch rack
¾
Supports 15 ports of tone detection
¾
Contains four media-module slots
¾
One Avaya P330 expansion-module slot
¾
One slot for the Avaya P330 Octaplane stacking fabric
¾
Can sit on a desktop or be rack-mounted.
¾
Contains an internal motherboard
¾
Standard based 10/100 Ethernet Interface connection types. A wall field
or breakout panel is not required.
¾
Internal power supply that provides low-voltage DC power to the fans,
motherboard, and media modules
¾
Four internal fans that provide cooling for the internal components
¾
A LED board that indicates system-level status
¾
A serial port for command-line access
¾
A VoIP engine that supports up to 64 G.711 single-channel calls
¾
Eight-port layer-2 switch
The G700 Media Gateway has a physical design that is similar to the Avaya
stackable switching products. This is the hardware that will provide the gateway
functionality at VoiceCon’s remote locations
Avaya G700 Media Gateway Architecture
Media Modules
Cajun Expansion Module
December 1, 2006
10/100 Base-T Ethernet Ports
©2006 Avaya Inc.
Serial CLI Connector
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
Avaya G700 Media
Requirements
Gateway
Form
Factor
and
Environmental
Chassis Dimensions
Height
2U (3.5
in)
88 mm
Width
19 in
482.6 mm
Depth
17.7 in
450 mm
Empty Weight
22.25 lbs
10 kg
Loaded Weight
27-34 lbs
12-16 kg
Required Clearances
Front
12 in
30 cm
Rear
18 in
45 cm
consistent with EIA 464 data rack standards
Temperature Tolerances
Recommended
65 to 85 degrees Fahrenheit
18 to 29 degrees
Celsius
+41 to +104 degrees
Fahrenheit
5 to 40 degrees Celsius
Continuous operation
Humidity Tolerances
Recommended
20 to 60% relative humidity
Relative humidity range
5% to 95% non-condensing
Altitude
Recommended – up to 10,000 feet or 3,000 meters
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
G250 Media Gateway
An enterprise branch office gateway designed to serve the communications needs
of a small branch with 2 to 14 extensions. This gateway allows organizations to
economically extend their headquarter communication applications such as the
Avaya Communication Manager and the Avaya Modular Messaging out to their
branch locations to achieve lower overall costs and increased collaboration across
their entire organization.
This is a powerful branch communication solution that packs an IP telephony
gateway, an advanced IP WAN router, and a high-performance LAN switch into a
compact, 2U high 19" rack mount unit. The G250 Gateway extends the enterprise
communications capabilities in the headquarters location out to a branch location
and is ideally suited for branch locations needing from 2 to 14 extensions. The
system gains its functionality from a centralized Avaya Media Server running Avaya
Communication Manager, but has several survivability options that allow
communications to continue operating even if the connection to the main server is
lost for any reason.
An advanced TDM/IP architecture provides seamless connectivity and
communications between a wide variety of analog, digital, H.323, and SIP-based IP
telephony devices and applications. To enhance security, the G250 can secure VoIP
media streams using Advanced Encryption Standard (AES), approved for use by
U.S. Government agencies to protect sensitive information.
The Avaya G250 also functions as an edge router to support the consolidation of
voice and data traffic over an IP network. Optional IP WAN routing media modules
add support for PPP/Frame Relay connectivity over E1/T1 or Universal Serial Port
(USP) interfaces. The G250 media gateway can also connect to an external WAN
device via a fixed 10/100 Ethernet WAN router port, which supports traffic shaping
to match data transfer rates with available WAN bandwidth.
Local Survivable Processor (LSP) Support
An Avaya S8300 Media Processor can optionally be installed in the G250 gateway
as a Local Survivable Processor to provide 100% Avaya Communication Manager
features if all connections to the central location is lost. As an optional feature, only
branch locations requiring this level of survivability need to have a Local Survivable
Processor installed.
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
Avaya G250 Media
Requirements
Gateway
Form
Factor
and
Environmental
Avaya G250 Media Gateway Dimensions
Width
17.3 in. (43.94 cm)
Height
3.5 in. (8.89 cm)
Depth
13.375 in. (33.97 cm)
Weight
22.0 lbs. (10 kg)
Avaya G250 Media Gateway – Environmental Specifications
BTUs
2866
Operating Temperature:
Recommended
32-104 deg F (0-40 deg C)
Operating Humidity: Recommended
95% non-condensing relative humidity
Operating Altitude
up to 10,000 feet or 3,048 meters
Minimum clearance for system cooling
Front 12 in. (30 cm)
Rear 18 in. (45 cm)
Consistent with EIA 464 data rack
standards
Power Rating
December 1, 2006
100-240 V~, 50-60 Hz, 2.2A <ax
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
IP Circuit Packs
IP
Server
TN2312BP)
¾
Interface
Card
(IPSI
Provides Control Interface for Media
Gateway
Control LAN Card (C-LAN TN799DP)
¾
Handles signaling for IP Phones and
Trunks
¾
Handles signaling for Adjuncts [Modular
Messaging, Call Management System
(CMS)]
¾
Delivers Tone and Call Classifications
Resources
¾
64MB flash memory and SDRAM
¾
Allows Remote Administration
¾
Board LEDs [power up (red),
maintenance (green), AA (amber), clock
(amber), Emergency Transfer (red) ]
¾
Dedicated resource, up to 450 sessions
¾
G650 Environmental Cabinet
Maintenance
¾
Emergency transfer
CLAN Card Illustration
IPSI Card Illustration
Media Processor Card (MedPro):
December 1, 2006
¾
Converts TDM based media in IP based
¾
Supports Codecs: G.711, G.729, G.723
¾
Supports from 32 to 64 simultaneous
sessions
¾
Dynamically allocated resource
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
Traditional Telephony Circuit Packs
Analog Stations and Analog Trunks
The G650 Media Gateway supports analog stations or trunks via the TN793B Analog
24 port Circuit Pack. It also uses the TN747B CO trunk circuit pack, which has eight
ports for loop- or ground-start CO, foreign exchange (FX), and wide area
telecommunications service (WATS) trunks. Each port has tip and ring signal leads.
A port can connect to a paging system. The TN747B supports the abandoned call
search feature in automatic call distribution (ACD) applications (if the CO has this
feature). Vintage 12 or greater of the TN747B circuit pack also provides batteryreversed signaling.
G700 and G250 Media Gateways support analog stations or trunks via the MM711,
8 port Media Module.
MM711 Analog Media Module
The MM711 provides the administrator with the capability to configure any of the
eight ports of this analog circuit pack as:
¾
A loop start or a ground start central office trunk
¾
Loop current 18-60mA
¾
A wink start or a immediate start Analog Direct Inward Dialing (DID)
trunk
¾
A 2-wire analog Outgoing CAMA E911 trunk, for connectivity to the PSTN
¾
MF signaling is supported for CAMA ports
¾
Analog, tip/ring devices such as single-line telephones with or without
LED message waiting indication
The MM711 also supports:
¾
Type 1 and Type 2 Caller ID
¾
Ring voltage generation for a variety of international frequencies and
cadences
¾
A hard-wired ground wire is added for each IROB-to-earth ground
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
MM716 Analog Media Module (Optional)
Optional Avaya MM716
24-port Analog Media Module
The MM716 Analog Media Module is a 24
dedicated for station (line). The station
Waiting Indicator, and are used for DID
connections. The MM716 is supported in
Communication Manager 3.0 or higher,
necessary to support the MM716.
port analog circuit pack. All ports are
(line) ports support an LED Message
trunk (wink or immediate start) port
the G350 and G700 Media Gateways.
and equivalent gateway software, is
Digital Trunks
The G650 Media Gateways use the TN464GP DS1 Interface, which supports T1 (24
channels) or E1 (32 channels). The interface provides:
¾
Board-level, administrable A- or µ-Law companding
¾
CRC-4 generation and checking (E1 only)
¾
Stratum-3 clock capability
¾
ISDN-PRI T1 or E1 connectivity
¾
Line-out (LO) and line-in (LI) signal leads (unpolarized, balanced pairs)
¾
Support for CO, TIE, DID, and off-premises station (OPS) port types that
use robbed-bit signaling protocol, proprietary bit-oriented signaling (BOS)
24th-channel signaling protocol, or DMI-BOS 24th-channel signaling
protocol
¾
Support for Russian incoming ANI
¾
Support for universal, digital, signal level-1equipment in wideband ISDNPRI applications
¾
Test-jack access to the DS1 or E1 line and support of the 120A integrated
channel-service unit (ICSU) module
¾
Support for the enhanced maintenance capabilities of the ICSU. These
circuit packs can communicate with Avaya CONVERSANT® or Avaya
Interactive Response.
¾
Downloadable firmware
¾
Support for echo cancellation
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
The echo cancellation capability of the TN464GP is selectable on a per-channel
basis. The TN464GP DS1 interface automatically turns off echo cancellation when it
detects a 2100-Hz phase-reversed tone generated by high-speed modems (56kbps), but not when it detects a 2100-Hz straight tone generated by low-speed
modems (9.6-kbps). Echo cancellation improves a low-speed data call.
The G700 and G350 Media Gateways use a MM710 E1/T1 Media Module to digital
trunks. The MM710 has a built-in Channel Service Unit (CSU) so that an external
CSU is not necessary. See the following illustration for an example of the MM710.
Avaya MM710 T1/E1 Media Module
Highlights of the MM710:
¾
Software selectable T1 or E1 operation
¾
An integrated CSU
¾
Both A-law (E1) and µ-law (T1) gain control and echo cancellation ability
¾
D4, ESF, or CEPT framing
¾
ISDN PRI capability (23B + D or 30B + D)
¾
AMI, ZCS, B8ZS (T1) or HDB3 (E1) line coding
¾
Trunk signaling to support US and international CO or tie trunks
¾
Echo cancellation in either direction
¾
Fractional T1 support
¾
An OIC DB 25-pin interface
¾
A Bantam loop back jack that is used for testing of T1 or E1 circuits.
The MM710 supports the universal DS1 that conforms to the ANSI T1.403 1.544
Mbps T1 standard and to the ITU-T G.703 2.048 Mbps E1 standard.
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
1.3.1 Common Control
The primary common control complex of the proposed IPTS should be based 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 set-up 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
Avaya Response:
Comply. The proposed common control is an Avaya S8720 Media Server stack. The
Avaya S8720 Media Server introduces a powerful processor, increased storage and
optical drive capacity, higher speed USB 2.0 interfaces, and a higher performance
server duplication interface, while retaining the 2U vertical form factor of the S8700
platform.
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
The Avaya S8720 Media Server stack includes two servers, one server is always
active the other is always in the standby mode. There is no reduction in the
feature/function capabilities if a back-up S8720 Media Server processor is activated
in case of a main call processor failure, and calls in process remain active.
The Avaya S8720 Media Server can support up to a system maximum of 15,424
simultaneous circuit switched calls or 242 per port network. There are 484 time
slots for voice and data per port network. For IP to IP calls this limitation does not
apply. The CCS rating for the S8720 Media Server is 600,000 general business type
Busy Hour Call Completions.
The S8720 Media Server is based on the powerful AMD Opteron processor with an
Enterprise Linux operating system. The S8720 Media Server with Communication
Manager 3.1 provides a rock solid foundation for a highly flexible converged
solution that meets a variety of telephony needs.
Avaya S8720 Media Server Specifications:
¾
¾
¾
¾
¾
¾
¾
Duplicated servers
19 inch rack mounted (3.38 in. (8.6 cm.) x 17.50 in. (44.5 cm.) x 25.75 in.)
2U form factor.
2.8 GHz AMD Opteron Processor
2.72 Gigabyte SCSI drive
1 Gigabyte RAM
Operating System – Linux Red Hat Enterprise Version 4.0
Memory
¾ Physical Standard Memory:1GB
¾ Optical Drives: DVD/CD-ROM
¾ NIC: 4 NICs (10/100) - 2 NICs (10/100/1000)
¾ Hard Drive: 1 72GB SCSI drive
¾ Slots: 3 PCI-X slots - 2 hot plug 64-bit/100MHz - 1 64-bit/133MHz
¾ Ports: 3 USB 1.1
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
Duplication
¾
Hardware and Software Options
o
S8720 Media Server with Hardware Duplication 2 - DAL1 duplication
boards (100MHz)
o
S8720 Media Server with Software Duplication No DAL1 boards needed.
Dimensions
¾
Width and Height Width and Height: 3.38 x 17.54 x 26.01 in (8.59 x 44.54 x
66.07 cm) per server
¾
Weight Approximate: 60lb (27.22 kg) per server
¾
Chassis: 2U (3.5") form factor per server
Electrical Requirements
¾
Voltage Rated Input Voltage: 100-132 VAC, 200-240 VAC
¾
Rated Input Current: 7.5A (100VAC), 3.8A (200VAC)
¾
Rated Input Frequency: 50 to 60 Hz
¾
Rated Input Power: 735 W
Operating Environment
¾
Temperature & Relative Humidity Temperature: 50 to 95 degrees F (10 to
35 degrees C)
¾
Relative Humidity: 10% to 90%
Capacity
¾
Station Support Station Support: 36,000 Total stations
¾
Trunks: 8000 Total Trunks
¾
IP Endpoints: 12,000 (cumulative total of IP trunks, IP stations and SIP
trunks)
¾
Supported Media Gateways (G250, G350 and G700): Up to 250
¾
Supported Port Networks - G650 - up to 64; SCC1/MCC1 - up to 64 for ATM
PNC and up to 44 for CSS
¾
Fax, Teletypewriter device (TTY), and modem calls over a corporate IP
intranet using pass-through mode
¾
T.38 Fax over the Internet (including endpoints connected to non-Avaya
systems)
Availability/Survivability
¾
Two levels of duplication; duplex and high
¾
Processors, control network and bearer network can be duplicated
December 1, 2006
©2006 Avaya Inc.
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
¾
Server separation up to 6 miles (10 Km)
1.3.2 CPU Make/Model
Vendor Response Requirement:
Identify the make/model of all proposed common control CPUs and associated BHCC
rating for the configured system.
Avaya Response:
The proposed system uses the Avaya S8720 Media Server with an AMD Opteron
processor with an Enterprise Linux operating system for common control. The
server supports busy hour calls as listed below.
¾
S8720 Media Server with Hardware Duplication 600,000 BHCC general call
type mix. (IP and Call Center impact performance.)
¾
S8720 Media Server with Software Duplication Approximately 250,000 BHCC
general call type mix. (IP and Call Center impact performance.)
1.3.3
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.
Vendor Response Requirement:
Briefly describe the main memory design and storage elements and capacity for
both the generic software and customer database as proposed.
Avaya Response:
Comply; The S8720 Media Server, a new version of the Avaya flagship S87XX
product line, reflects our core Media Server strategy to deliver on the
price/performance promise of the Avaya Communication Architecture by regularly
updating our Media Server platforms to take advantage of new technologies and
performance improvements.
The S8720 Media Server is based on the powerful AMD Opteron processor with an
Enterprise Linux operating system. The S8720 Media Server with Communication
Manager 3.1 provides a rock solid foundation for a highly flexible converged
solution that meets a variety of telephony needs.
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The S8720 Media Server is always sold with two duplicated servers to ensure
maximum up time. The S8720 Media Server will be available in two configurations:
with Hardware Duplication (requiring the two DAL1 duplication boards) or with
Software Duplication (no DAL1 boards needed). With Hardware Duplication, the
S8720 Media Servers can be separated up to 10 kilometers (6.3 miles) to help
ensure business continuity. With software duplication the S8720 Media Server
separation distance is governed by the capacity and quality of the duplication link.
For software duplication to function properly, the minimum network requirements
for the duplication link are:
¾ 1 GB Ethernet link, minimum
¾ 8 ms round trip delay, maximum
¾ 0.15% roundtrip packet loss, maximum
The S8720 Media Server, with hardware duplication (Proposed), can process up to
600,000 Busy Hour Call Completions (BHCC) in a general call type mix. The BHCC
for the S8720 Media Server with Software Duplication is approximately 250,000 in
a general call type mix.
Connecting the Cables: Duplication Cabling for Hardware Duplication
The S8720 Media Server can support up to 36,000 stations and 44,000 ports, up to
12,000 IP endpoints (which is a cumulative total of IP trunks, IP stations and SIP
trunks), and 8,000 trunks enabling it to support large multi-national corporations
and contact center operations.
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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?
Avaya Response:
To protect the integrity of the data, the "save translation" command writes two
time-stamped identical copies of the configuration data to the selected memory
card, disk, or tape. The save writes one complete copy first, and then writes the
second copy in a different area of the device — both with the same time-stamp.
Failure during a save, including a system crash, usually affects only one copy. The
affected copy is marked "bad" and should not be used for backup.
Best practice would recommend off-site storage for all valuable company
databases, including the telephony server database. In addition to daily
maintenance, the Web Services inherent in any Avaya Linux-based Media Server
can FTP (or email) the translation database to an off-site server (preferably at a
hardened location with RAID drives) on a daily basis.
1.3.4.2 Database Information Loss
Vendor Response Requirement:
Identify under what circumstances can customer database information (configuration,
messages, logs, etc.) be lost during back-ups
Avaya Response:
The standard daily maintenance cycle would save database changes / translations
to an Avaya Linux-based Media Server hard drive, and to any Enterprise Survivable
Servers (ESS) and Local Survivable Processors (LSP) hard drives in your network.
Therefore move, add, and change activities completed for a particular day are
saved to the Avaya Media Server’s hard drive and distributed to ESS and LSP hard
drives automatically.
Any changes made between daily maintenance cycles would be lost unless the
administrator uses the “save translation” command
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1.3.4.3 Database Backup Scheduling
Vendor Response Requirement:
How often should the customer database be backed up? Specify if it is a full or
incremental backup and the time the process takes.
Avaya Response:
One of the most important administrative tasks is to set up a schedule to routinely
back up system data to a safe location. While the detailed procedure is up to Voice
Con, Avaya supports multiple backup options. A backup procedure can run
manually or automatically based on a schedule created by your administrator.
Possible destinations for the backup files are an FTP server, email, and local PC
card. The web interface allows the administrator to back up call-processing data
(Avaya Communication Manager “translations”), server system data, and security
files. Avaya recommends that the system administrator encrypt the backup files to
keep this sensitive information secure.
Administrators can use a browser with SSH to access the server and schedule
backups, as well as restore the configurations. Alternately, a locally connected
console can be used to store/restore the configurations.
We recommend that backups be done at least once per day (especially of the
translation set). The backups are “full”; i.e. they contain all of the data specified in
the set, not just the modified files. Typically, a backup set for Communication
Manager sets will take on average about 3 minutes or less. However, this is an
average value, because the “translation” set can take longer dependent on the size
of the translation file, and if a save translation has been requested.
1.3.4.4 Data Purging/Archiving
Vendor Response Requirement:
Describe the mechanism for data purging and archival, including storage and retrieval of
archived data.
Avaya Response:
The storage of the images are totally the customer’s responsibility. That is we
provide the mechanism to create the images; but the customer determines where
the images are to be retained (remote server via ftp, etc, or compact flash). We do
not support or recommend storing the images to the local server, for the mere fact
that if the servers become inaccessible, the customer will no longer be able to
access his archives
The user has the ability to see the archive information for the machine via the web
menus or they can issue a query directly to the destination device/system; in this
particular instance, the query will display a list of archival files that are at the
specified destination.
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None of the images are ever overwritten by avaya. Each image has a unique
date/time stamp as a part of the image name; therefore the files will remain at the
destination server until removed by VoiceCon. In the case of Compact Flash
destinations, the user has the option to specify how many copies of each archive
type they want to be kept on the system (up to a maximum of 9 copies, It will
always be the latest 9 backup images).
1.3.5 Power Supply
Vendor Response Requirement:
Briefly describe common control power requirements and the integrated power
distribution design. Indicate if the power supply is dependent on either an AC or DC
current source.
Avaya Response:
The Avaya S8720 Media Server global power ranges from Support for global power
ranges from 100V to 250V AC.
The G650 Media Gateway has 2 AC Inputs, 1 DC Input, as described below:
¾ AC: 100-120 VAC at 50Hz – 60 Hz, 9.0 Amps Max
¾ AC: 200-240 VAC at 50Hz – 60 Hz, 4.5 Amps Max
¾ DC: -48 VDC (-40 VDC to –60 VDC), 21.0 Amps Max
The G700 Media Gateway uses an auto-ranging 100-240 Vac power supply, 50 to
60 Hz, 5 A maximum at 100-120 Vac and 2 A maximum at 200-240 Vac. The AC
power source is to be single phase, 3-conductor (Line, Neutral and Ground) with a
15 A circuit breaker for 100-120 Vac or a 10 A circuit breaker for 200-240 Vac.
The G250 Media Gateway requires 100-240V~, 2.2 A Max, 50-60Hz
1.3.5.1 Power Safeguards
Vendor Response Requirement:
Describe any power failure safeguards that are included in the IPTS design. Briefly
describe what happens to system operation during a power failure
Avaya Response:
The Avaya S8700-Series Media Server has duplicate processors. If an active Media
Server fails, and the backup (standby) Media Server takes over, it is a “stateful
failover” process, meaning that calls and all the characteristics of the calls are
maintained. Any failures at this level are completely transparent to all end users
and devices. No re-registration is necessary.
During a complete power failure at the HQ site the power failure transfer equipment
will activate allowing access to local two-way CO trunks via VoiceCon provided
analog telephones. At that point, the RO site will no longer be able to
“communicate” and will activate its ESS mode allowing any IP telephones or
“powered up” media gateways to register as defined in the translations.
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IP Telephones and media gateways receive a list of alternate gatekeepers as part of
the standard configuration. If the endpoints cannot reach their primary gatekeeper,
the endpoints will re-register with the secondary gatekeeper. If they cannot reach
that gatekeeper, they will attempt to register with the tertiary gatekeeper (an LSP
at the Small Branches), and so on.
RELIABILITY AND PERFORMANCE
One of the strengths of the Avaya Media Server Architecture is the reliability
options available. Those options include simplex reliability (S8500), duplex
reliability (S87XX), high reliability (S87XX), and critical reliability (S87XX).
Each of these options builds on duplication of critical components. Simplex
reliability is available with S8500 configurations and utilizes a single server. Duplex
reliability duplicates the S87XX Servers. The S8720 Media Server is always sold
with two duplicated servers to ensure maximum up time. The S8720 Media Server
is available in two configurations: with Hardware Duplication as presented in this
design (requiring the two DAL1 duplication boards) or with Software Duplication (no
DAL1 boards needed). With Hardware Duplication, the S8720 Media Servers can be
separated up to 10 kilometers (6.3 miles) to help ensure business continuity. With
software duplication the S8720 Media Server separation distance is governed by the
capacity and quality of the duplication link. For software duplication to function
properly, the minimum network requirements for the duplication link are:
¾
1 GB Ethernet link, minimum
¾
8 ms round trip delay, maximum
¾
0.15% roundtrip packet loss, maximum
High reliability adds duplication of the control network and its component parts,
including Ethernet switches, duplicated control network interfaces on the S87XX
Servers, and duplicated IP Server Interfaces (IPSI) in the Port Networks. Critical
reliability adds duplication of the bearer with duplicated IP Media Resource boards.
When all of these components are duplicated, five nines of reliability (99.999%) can
be obtained in the components associated with the system.
Business Continuity
Communication Manager powered solutions can achieve up to 99.999% reliability
through the use of duplicated servers interfaces and network links. The Avaya
Enterprise Survivable Server (ESS) solution allows business the flexibility of
consolidation by providing increased survivability options. In addition, a variety of
new enhancements like Inter-Gateway Alternate Routing (IGAR), Locally Sourced
Announcements & Music, Auto Fallback to Primary for H.248 Gateways, Modem
Dial-up the G350 and G250 media gateways, Standard Local Survivability on the
G250 media gateways and Connection Preserving Failover/Failback for H.248 Media
Gateways can be used to provide customers at the branch office uninterrupted
communication service.
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The combination of new Communication Manager features with G250 Media
Gateway enhancements and new high-capacity DCP media module for the
G250/G350/G700 media gateways provide a very compelling offering for the
growing distributed branch office market. LSP capacity now matches H.248 media
gateway capacity, which means branch office deployments, can now be fully
deployed with LSP support. To support this increased capacity, a new LSP
translation synchronization feature improves bandwidth utilization for file updates.
Enterprise Survivable Server
Enterprise Survivable Servers allow backup servers to be placed at various places in
a network so that communications can continue in the event that a main server(s)
fails, or when connectivity to the main server(s) is lost. Enterprise Survivable
Servers can be connected to ATM, IP and Center Stage Switch (CSS) connected
port networks. The IPSI card in the port networks automatically obtains service
from an ESS server(s) if the control signal to the main server is lost. In Avaya
Communication Manager (CM) 3.1, there can be up to 63 Enterprise Wide ESS
servers in a network.
Deploying Enterprise Survivable Servers at appropriate points in a network provides
flexibility in how to plan against system failover. Assurance of communications
continuity can be mapped to protect against network failures or catastrophic main
server failures or both. Administration for the ESS can be conducted in a central
point with automatic synchronization to all ESS servers.
1.3.5.2 Power Backup
Vendor Response Requirement:
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?
Avaya Response:
The Media Server pair is always recommended with power backup for each of the
servers. Power backup is required to avoid power problems and to ensure graceful
shutdown of the system processes if the power fails. Combinations of battery
extension modules and a 1500-VA UPS can provide up to eight hours of power
backup.
Avaya considers voice communications to be mission-critical to customers'
enterprises, and strongly recommends the use of an online Uninterruptible Power
Supply (UPS) with each S8720 Media Server. Customers may provide a UPS to
support each S8720 Media Server with a 400VA and 3.33AMP rating for each
server.
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1.3.6 Ethernet Interface Call Control Signaling Links
Vendor Response Requirement:
Identify for each active and standby call processor element the number of available and
configured RJ-45 Ethernet LAN uplink interfaces for call control signaling to LANconnected cabinets/carriers and/or standalone ports. Include a brief description of how
the physical Ethernet connection is provided: dedicated circuit board; daughterboard;
fully integrated RJ-45 connector, et al.
Avaya Response:
The Avaya Media Servers provide integrated RJ-45 connectors. A connection is
established between the Media Servers and the customer-provided layer-2 devices.
From the data switches, a connection is established to a dedicated circuit pack, the
Internet Protocol Server Interface (IPSI-2). This signaling network is duplicated in
the Avaya S8720 Media Server, with a Control Network A (CN-A) and Control
Network B (CN-B) terminating on duplicated IPSI-2.
In the case of the Avaya G700 Media Gateways, control signaling begins at the
Media Servers, traverses the control network through the G650 Media Gateway,
through the Control LAN (C-LAN), terminating at the G700 and G350 Media
Gateway on a integrated RJ-45 port. VoiceCon’s data infrastructure provides the
transport between the C-LAN and G700 and G250, depending on the configuration.
As with the G700 and G250 Media Gateways, all IP endpoints terminate signaling
traffic on the C-LAN gatekeeper, before continuing on to the Media Server. Please
refer to the included diagrams for a more detailed description of the physical
connectivity.
Avaya S8720 Media Server Signaling Link Connectivity
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1.3.7 System Clocks
Vendor Response Requirement:
Identify the number and type of internal system clocks that are available and configured.
Avaya Response:
Tone/Clock functions are provided by the IP System Interface (IPSI – TN2312BP)
board. The TN2312BP generates tones and provides clock functions for the port
network in which it is placed. This functionality is equivalent to the TN2182B
Tone/Clock circuit pack.
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:
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.
z
Primary call processor
Avaya Response:
Comply. The proposed S8720 Media Server Stack is a duplicated back-up solution,
with seamless (call and connection preserving) switchover.
Avaya has proposed duplex reliability for VoiceCon’s solution of ESS and LSP
backup servers the optional high reliability configuration could be achieved by
adding a few optional pieces of equipment which will consisting of the following:
The high-reliability configuration option builds on the duplex-reliability option. The
high-reliability option duplicates components so that no single point of failure exists
in the control network. The high reliability configuration consists of the following:
¾
Two S8720 Media Servers (included in duplex reliability design)
¾
Two Ethernet switches (Provided by the customer’s network)
¾
Two UPS units (Customer Provided)
¾
Two IPSI circuit packs in each IPSI-connected port network (provided with
design)
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1. S8720 Media Server pair. One in active mode and the other on standby.
2. Duplication Links: The Ethernet connection default Ethernet 2 and the fiber
link
3. A dedicated Ethernet connection to a laptop. This connection is active only
during on-site administration or maintenance and the services interface can
link to the standby server through a telnet session.
4. Connection from the servers to the Ethernet switch.
5. Ethernet Switch – A device that provides port multiplication on a LAN by
creating more than one network segment. In an IP-Connect environment, the
Ethernet switch should support 802.1 ip/Q, VLAN and 10-/100-Mbps.
6. Two UPS units (Customer Provided)
7. Port Network – An optional configuration of Media Gateways that provides
increased port capacity.
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8. IPSI – A circuit pack that transports control messages over IP. This IPSI
circuit pack is used so the S8720 Media Server can communicate with the
PNs.
9. Customer LAN
10.Control LAN Interface (C-LAN) - A circuit pack that provides call control for
every IP endpoint that is connected to the Media Server using an Avaya
media gateway.
z
Main system memory
Avaya Response:
The main system memory is fully redundant, based on the duplicated server design.
z Customer database memory
Avaya Response:
Using a patented memory shadowing technique, each of the redundant S8720
Media Servers has a complete customer database stored at all times.
z RJ-45 Ethernet uplinks to network
Avaya Response:
Ethernet uplinks are fully redundant and supported by separate Voice Con provided
LAN switches.
z Power supply
Avaya Response:
Power is fully redundant; each server has a dedicated power supply and a VoiceCon
provided dedicated UPS per server.
z Tone generators
Avaya Response:
Tone generation is load sharing, The IPSI provides this functionality as well as two
TN744E call classifier and tone detector packs provided in the HQ location, 1 in the
RO. These are built into the small branch gateways
z Call classifiers
Avaya Response:
Tone generation is load sharing, The IPSI provides this functionality as well as two
TN744E call classifier and tone detector packs provided in the HQ location, 1 in the
RO. These are built into the small branch gateways
z Registers
Avaya Response:
Comply. The IPSI provides this functionality
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z DTMF receivers
Avaya Response:
The TN744 circuit pack supports digital signal processing of PCM signals on each
port to detect, recognize, and classify tones and other signals. Generation of
signaling tones is also supported for applications such as R2-multifrequency code
(R2-MFC), Spain MF, and Russia MF. Gain or loss and conferencing can be applied
to PCM signals that are received from the TDM bus. Additional support includes
DTMF detectors to collect address digits during dialing, and A- and μ-Law
companding
z I/O interfaces
Avaya Response:
VoiceCon is providing the LAN Switches connected to the S8720 Media Server;
however for the High Reliability Option the solution requires redundant LAN
switches (I/O Ports).
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 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.
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Avaya Response:
Comply. The proposed solution includes an S8720 Media Server stack (duplicated
servers) at the HQ location and an ESS server at the Regional Office. The in the
event the WAN is lost to HQ and there is a connection available to the RO all SB
offices would re-register to the RO facility and continue operations as normal.
The SBs will have a Local Survivable Processor (LSP) at SB1 and SB2 branch
locations, the very small branch will use the local survivability however the IP
phones at that location can be translated to reregister to the RO or another LSP
location. In the very Small Branch (SB3) an LSP can be added if desired. In the
event the RO is unreachable, the LSP would activate. The LSP is an S8300 Media
Server is a blade housed within the Avaya H.248 gateway (G250 or G700) at each
branch. The LSP maintains a copy of the S8720 Media Server customer
translations. An automatic process copies customer translation changes made on
the primary server to every LSP in the network. The translations are updated
regularly using a virtual link via the IP network. To maximize effectiveness while
conserving bandwidth, a file difference mechanism identifies changed records within
the translation file and transmits only changed records when file synchronization
occurs.
S8300B Media Server Module
Typically, all ESSs/LSPs are in idle mode, where the ESS/LSP is not processing any
calls. When the Media Gateway’s Processor (MGP) or IP endpoints perceive the
Avaya Media Server to be unreachable, the MGP or IP endpoints will attempt to
register with a predefined ESS/LSP. The LSP does not actively take over when the
primary controller becomes unreachable, but waits for MGPs and IP endpoints to
register with it. If for any reason communication between an Avaya H.248 Media
Gateway and its primary controller stops, the LSP activates. This "fail-over" from
the primary controller to the ESS/LSP is an automatic process without human
intervention. The ESS/LSP assumes control of any IP telephone provided that
telephone has the ESS/LSP in its list of controllers. A feature called ConnectionPreserving Failover/Failback for H.248 Media Gateways provides uninterrupted
communication service for stable calls at the branch office when switching from an
active/main component to a standby/backup component or vise versa. All features
and capabilities supported by Avaya Communication Manager are available to users
at each site when in survivable mode.
IP telephones obtain their own IP address from a DHCP server. The DHCP server
also sends a list of controllers, LSPs, and their associated IP addresses. The IP
telephone then registers with the controller corresponding to the first IP address in
this list. When connectivity is lost between the controller and the endpoint, the
endpoint registers with the second IP address in the list, and so on. This list is
administrable for telephones on the DHCP server.
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For each location, PSTN trunks were requested per your specifications, so the
requested PSTN facilities will support the location in case of a WAN failure.
1.4.1
Survivable IPTS Services
Vendor Response Requirements
Identify any required generic software feature (See Section 5.0 Call 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.
Avaya Response:
The Connection Preserving Migration (CPM) feature preserves existing bearer
(voice) connections while an H.248 media gateway migrates from one
Communication Manager Media Server to another. Migration might be caused by a
network or server failure. Only stable calls are preserved. Calls that are not
preserved during the transition to ESS/LSP mode are:
¾
Unstable calls. An unstable is any call where the call talk path between
parties has not been established, or is not currently established. Some
examples are:
o
Calls with dial tone
o
Calls in dialing stage
o
Calls in ringing stage
o
Calls listening to announcements
o
Calls listening to music
o
Calls on hold (soft, hard)
o
Calls in ACD queues
o
Calls in vector processing
o
IP trunks, both SIP and H.323
o
ISDN-BRI telephones
o
ISDN-BRI trunks
Users on connection-preserved calls cannot use such features as Hold, Conference,
or Transfer.
Refer to response in 1.4 above for additional information
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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?
Avaya Response:
Comply. A feature of Avaya Communication Manager called Auto Fallback to
Primary for H.248 Media Gateways automatically returns a fragmented network,
where a number of H.248 media gateways are being serviced by one or more Local
Survivable Processors (LSP), to the primary Media Server.
Auto fallback to primary for H.248 Media Gateways allows your administrator to
define any of the following rules for migration/fallback to the main server:
¾
Whether or not the media gateways, serviced by ESS/LSPs, should
automatically migrate to the primary media gateway.
¾
Whether or not the media gateway should migrate immediately when
possible, regardless of active call count.
¾
Whether or not the media gateway should only migrate if the active call
count is 0.
¾
Whether or not the media gateway should only be allowed to migrate within
a window of opportunity, by providing day of the week and time intervals per
day. This option does not take call count into consideration.
¾
Whether or not the media gateway should be migrated within a window of
opportunity by providing day of the week and time of day, or immediately if
the call count reaches 0. Both rules are active at the same time.
Internally, the primary call controller gives priority to registration requests from
those media gateways that are currently not being serviced by an LSP
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1.4.3 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.
Avaya Response:
The proposed messaging solution is centralized at the HQ location with access to
the branch location(s) via the IP connections. In the event the IP Connection (WAN)
fails, callers can still leave messages at the central messaging system. End user’s
can access their messages via the TUI by dialing through the PSTN. This would
preclude them receiving message waiting and any GUI access to messages. The
minimum messaging requirements is met in the recommended configuration.
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.
Avaya Response:
Comply
The ESS positioned at the RO facility will support all licensed ports from the HQ
location
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The Avaya S8700-Series Media Server has duplicate processors. If an active Media
Server fails, and the backup (standby) Media Server takes over, it is a “stateful
failover” process, meaning that calls and all the characteristics of the calls are
maintained. Any failures at this level are completely transparent to all end users
and devices. No re-registration is necessary.
Also included in this design for the SB1 & SB2 locations optional on the G250 at the
SB3 is a requested Local Survivable Processor (LSP) back-up server solution.
Typical need for a back-up server in an Avaya world would be on remote locations,
where the loss of connectivity isolates a remote office. For these situations, Avaya
offers two options. The automatic S8300 Local Survivable Processor (LSP) (SB
Locations) that does not require any human intervention to provide Business
Continuity for up to 450 users provides service to multiple locations and media
gateways (G700) if connectivity allows, and supports full functionality and database
transparency from the main site. To take this a step farther, a spare pair of S8700Series servers can also be configured where with human intervention; you can
restore the spare pair of S8700-Series servers into service and provide a speedy
response to disasters.
An additional option for survivability is an Avaya Enterprise Survivable Server
(ESS), which is provided as requested in this design at the RO facility, which allows
an enterprise like VoiceCon using a current Avaya Communication Manager driven
solution to have greater flexibility of consolidation by providing alternative
survivability options. The ESS solution allows VoiceCon to place ESS (backup)
servers at key locations throughout your enterprise. An ESS server is capable of
taking over for the entire enterprise or, if needed, just a portion of the enterprise in
the event that some type of outage has occurred. An enterprise typically involves
multiple locations that are equipped with Avaya port network gateways (G650)
and/or H.248 media gateways (G700, G250). These gateways use the Wide Area
Network (WAN) in order to communicate with the main Avaya Server. A port
network gateway uses a board called the Internet Protocol Server Interface (IPSI)
to establish connectivity to the Main server. The IPSI has the ability to "ask for
help" if connectivity to the mains server cannot be restored in a defined timeframe.
Each IPSI keeps a list of all the available ESS servers in the enterprise. The IPSI
will connect with the ESS server that has the highest priority rating to request
service. If that request for service fails, then the IPSI will make a request of the
next highest priority server.
Deployment of Enterprise Survivable Servers in your network can offer several
benefits to VoiceCon, including increased flexibility in how the system fails over.
You can choose to protect against network failures, catastrophic main server
failures, or both.
All Avaya survivability solutions are easy to administer, translations are completed
at one central point (HQ) and automatically synchronized throughout the network.
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IP Telephones and media gateways receive a list of alternate gatekeepers as part of
the standard configuration. If the endpoints cannot reach their primary gatekeeper,
the endpoints will re-register with the secondary gatekeeper. If they cannot reach
that gatekeeper, they will attempt to register with the tertiary gatekeeper, and so
on.
Avaya provides multiple signaling channels by the deployment of C-LAN cards. This
is one of the most important competitive advantages in the Avaya architecture.
End-points and remote Media Gateways (G700) register with one of the multiple CLAN cards in order to get telephony services. Moves, Adds, and Changes are done
that way as well. Because the endpoints and computers talk to one of those
multiple signaling channels instead of talking directly to the servers, customers can
isolate the S8700 series servers in their own network that can be physically
independent from the customer's network or even logically isolated in its own VLAN
using a non-routable "Secret Network", where nobody needs to know their IP
address. This provides awesome protection for servers, dispensing the need for
virus updates and pretty much eliminating the risk of Denial of Service (DoS) and
hacker attacks. By having multiple redundant C-LANs, we can connect them to
multiple different blades on the core Ethernet switches or to different switches,
providing resiliency towards network segment failure and DoS attacks. If attacked,
the C-LAN recovers shortly after the attack is done. If you lose a C-LAN, the
endpoints can automatically look for an alternate one for service. Load sharing is
done automatically between all the C-LANs in a network region.
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. SIP capabilities should conform, at minimum,
to RFC 3261, 1889, and 28331
Avaya Response:
Comply. Avaya offers SIP endpoints and SIP trunking via the Optional SIP Enabled
Services (SES) that is deployed in conjunction with Avaya Communication Manager,
and all SIP telephony users are administered in the SES and configured as
Outboard Proxy SIP (OPS) stations within Communication Manager. Adding an
Avaya SES to your enterprise enables the following:
¾
1
Avaya Communication Manager becomes a telephony feature server accessible
from a SIP endpoint – allowing access to the rich Avaya Communication
Manager feature set to support capabilities not provided by SIP
Added per e-mail from Alan on Tuesday, October 17, 2006 10:51 AM
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¾
Enterprise Instant Messaging (IM) functionality – a component of Avaya IP
Softphone Release 5. Think of this as an advanced IP softphone application with
an IM client that integrates a buddy list of voice and IM communication. IM
messages are encrypted and can be achieved to meet corporate communications
standards
¾
Our SIP telephones offer the standard telephony features of SIP and the value
add features of Communication Manager. These are the same IP telephones that
are proposed, but with a different load of firmware (SIP vs. H.323 load)
¾
SIP trunking for carrying media streams over VPN or SIP service provider media
gateways
¾
SIP integrates the communication silos of your voice portal (office phone and
cell phone) and text messages (IM and email) into one logical address to reach a
user
¾
Improves productivity by reducing phone tag with "presence awareness"
¾
Interoperable products from multiple vendors will drive the price down; in fact
there are currently SIP telephones available now for under $100.
The Avaya implementation allows our customers to implement SIP functionality
while still providing the robust features of Communication Manager that the SIP
standard does not address.
The Avaya implementation will allow any Avaya Communication Manager endpoint
(DCP, analog, IP) to be addressable with a SIP URL to protect most your initial
investment
Avaya SIP Enablement Services (SES)
Avaya provides an optional standards-based SIP architecture for telephony, Instant
Messaging, and other enterprise communications. SES extends the power of Avaya
Communication Manager to SIP telephony endpoints, and works with Avaya IP
Softphone to introduce presence-based telephony and instant messaging integrated
over a common user interface. The SES runs over Linux on one or more Avaya
Media Server platforms (with additional memory), enabling you to integrate SIPbased and conventional telephony.
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Avaya Communication Manager supports the following RFCs:
¾
RFC 791 – The G250 and G350 Media Gateways support IP fragmentation
and reassembly The G250/G350 router can fragment and reassemble IP
packets according to RFC 791. This feature allows the router to send and
receive large IP packets where the underlying data link protocol
constrains the Maximum Transport Unit (MTU).
¾
RFC 792 – Internet Control Message Protocol (ICMP), an extension to the
Internet Protocol (IP) defined by RFC 792. ICMP supports packets
containing error, control, and informational messages
¾
RFC 793 – Transmission Control Protocol (TCP), a connection-oriented
transport-layer protocol, RFC 793, which governs the exchange of
sequential data. Whereas the IP deals only with packets, TCP enables two
hosts to establish a connection and exchange streams of data.
¾
RFC 951 – Protocol that defines BOOTP, which allows the workstation to
boot without requiring a hard disk or diskette drive. It is used when the
user or station location changes frequently.
¾
RFC 951, 1534, 1542, 2131, and 2132 – Dynamic Host Configuration
Protocol (DHCP), an IETF protocol (RFCs 951, 1534, 1542, 2131, and
2132) that assigns IP addresses dynamically from a pool of addresses
instead of statically. DHCP provides the IP address to the IP devices, both
H.323 and SIP.
¾
RFC 1034, RFC 1035, and RFC 1123 – The DNS Resolver in the G250 and
G350 Media Gateways is fully compliant with RFC 1034, RFC 1035, and
RFC 1123
¾
RFC 1058 – Defines RIPv1, which is the original version of the RIP routing
protocol
¾
RFC 1213 – The SIP Enablement Server (SES) supports standard SNMP
MIB-II
¾
RFC 1490 and RFC 2427 – Frame Relay encapsulation
¾
RFC 1889 and RFC 3550 – Real-time Transfer Protocol (RTP), an IETF
protocol (RFC 1889 and 3550) that addresses the problems that occur
when video and other exchanges with real-time properties are delivered
over a LAN that is designed for data. RTP gives higher priority to video
and other real-time interactive exchanges than to connectionless data.
¾
RFC 2138 – Remote Authentication Dial In User Service (RADIUS)
¾
RFC 2198 – RTP Payload for Redundant Audio Data (that is, packet
redundancy)
¾
RFC 2246 – Transport Layer Security (TLS), which is an IETF standard
(RFC 2246) that supersedes Netscape’s’ Secure Socket Layer (SSL) and
provides a host-to-host data connections with encryption and certification
at the transport layer.
¾
RFC 2474 and RFC 2475 – Differentiated Services (DiffServ) complies with
RFC 2474 and RFC 2475
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¾
RFC 2401 to RFC 2412 – The G250 and G350 Media Gateways VPN
capabilities support Standards-based IPSec implementation (RFC. 2401 to
RFC 2412)
¾
RFC 2453 – Defines RIPv2, which is a newer version of the RIP routing
protocol
¾
RFC 2507 and RFC 1144 – The Avaya G250 and G350 Media Gateways
offer two options for configuring header compression:
¾
IP Header Compression (IPHC) method, as defined by RFC 2507, and
applies to RTP, TCP, and UDP headers
Van Jacobson (VJ) method, as defined in RFC 1144, and applies to TCP headers
only.
¾
RFC 2833 – Avaya's TTY over IP (TTYoIP) support works by identifying
TTY Baudot tones at the near-end Media Processor, removing them from
the voice path, and transporting them across the network in RFC 2833
messages. The far-end Media Processor receives the RFC 2833 messages
and regenerates them for the far-end station.
¾
RFC 3164 – The Avaya G250 and G350 Media Gateways include a logging
package that collects system messages in several output types. Syslog
Logging messages are sent to up to three configured servers, using
Syslog protocol as defined in RFC 3164. Messages sent to the Syslog
server are sent as UDP messages.
¾
SIP is built around published standards. These standards include both
IETF Requests for comments (RFCs) and Internet-Drafts. The standards
that the Avaya SIP solution implements include, but are not limited to,
these standards:
o
RFC 3261 (SIP)
o
RFC 3265 (SIP Event Notification)
o
RFC 3515 (SIP REFER Method)
o
RFC 3842 (SIP Message Summary and Message Waiting Indication
Event Package)
o
RFC 2327 (Session Description Protocol)
o
RFC 3264 (SDP Offer/Answer Model)
o
RFC 2617 (HTTP Digest Authentication)
o
RFC 3325, "Network Asserted Identity" is complied with on the SES
proxy servers
o
RFC 3891, "The SIP ’Replaces’ Header"
o
RFC 4028, "Session Timers in the SIP"
SIP endpoint (station and trunk) support in Avaya Communication Manager
complies with SIP standards, specifically IETF RFC 3261, and so interoperates with
any SIP-enabled endpoint/station that also complies with the standard.
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1.5.1
SIP Stations
Vendor Response Requirements
Indicate if the IPTS solution as proposed can currently support SIP-compatible 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.
Avaya Response:
Comply. Avaya has tested at SIPit with numerous third party SIP telephones,
including Alcatel, AT&T, and Cisco.
Avaya SIP Enablement Services are fully compliant with RFC 3261 and other core
SIP standards.
Enterprises can leverage SIP-enabled Avaya MultiVantage and third party
communication applications as multi-vendor services available over an open
architecture to a wide range of devices, user agents, and business applications.
SIP Endpoints. Gateways, Session Border Controllers and Security Products from
numerous third party vendors are supported. The following SIP standards based IP
endpoints have been verified as interoperable: The associated links take you to the
application notes.
Snom –
http://www.avaya.com/master-usa/enus/resource/assets/applicationnotes/snomsip.pdf
Counterpath eyeBeam SIP SoftPhone –
http://www.avaya.com/master-usa/enus/resource/assets/applicationnotes/eyebeam.pdf
Cisco 7940 and 7960 SIP Telephones
http://www.avaya.com/master-usa/enus/resource/assets/applicationnotes/cisco7960sipc.pdf
Blackberry Wireless Handheld
http://www.avaya.com/gcm/master-usa/ens/resource/assets/applicationnotes/avayabb7270.pdf
Hitachi Cable WIP-5000
https://devconnect.avaya.com/public/download/dyn/WIP5000RAD.pdf
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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?
Avaya Response:
Comply.
The 9600 Series telephones can be configured to support either H.323 protocol, for
integration with traditional Avaya IP Telephony Servers and Gateways, or SIP
protocol, for support of SIP Communications Servers such as the Avaya SIP
Enablement Services (SES) Release 3.0 Solution. The signaling protocol can be
toggled from the dial pad via a local administrative code (after which the
instrument will acquire the alternative firmware from the local HTTPS/HTTP server).
Avaya SIP Enablement Services (SES) creates a communication services layer
within the Avaya Communications Architecture that mediates between Avaya
MultiVantage applications and a wide range of standards-based user agents, webbased applications, and communication devices. These services combine the
standard functions of a SIP proxy/registrar server with SIP trunking support and
duplicated server features to create a highly scalable, highly reliable SIP
communications network. This resulting network supports telephony, instant
messaging, conferencing, and collaboration solutions.
Avaya SIP Enablement Services provide a compelling value proposition that allows
enterprises to maximize economic and productivity gains through the
implementation of SIP interoperability and capabilities, with an evolutionary path to
migration that fully protects existing investments and minimizes service and
business interruption.
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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.
Avaya Response:
SIP call control requires the SES server.
Avaya used a commercial SIP framework as basis for development. We participate
in SIPit (bi-annual SIP interop convention) to test our SIP capabilities for basic
calls, presence and instant messaging. We have tested at SIPit with Alcatel, AT&T,
AudioCodes, AudioTest, Broadcom, Cisco, Compaq, Digaco, dynamicsoft, HCL,
Hughes, Indigo, Mediatrix, Mitel, NIST, Nokia, Nuera, Pingtel, Polycom, Radcom,
Radvision, Siemens, SNOM, Sylantro, Telogy, TonesTest, Trillium, Vovida, Webley,
Wipro, and Worldcom. Avaya leverages standard test tools in development (e.g.
Navtel) and is pursuing independent certification (e.g. MierComm). Additionally
Avaya will work on interoperability with selected partners (e.g. Level (3))
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.
Avaya Response:
Comply.
Avaya SIP Enablement Services (SES) creates a communication services layer
within the Avaya Communications Architecture that mediates between Avaya
MultiVantage applications and a wide range of standards-based user agents, webbased applications, and communication devices. These services combine the
standard functions of a SIP proxy/registrar server with SIP trunking support and
duplicated server features to create a highly scalable, highly reliable SIP
communications network. This resulting network supports telephony, instant
messaging, conferencing, and collaboration solutions.
Avaya SIP Enablement Services provide a compelling value proposition that allows
enterprises to maximize economic and productivity gains through the
implementation of SIP interoperability and capabilities, with an evolutionary path to
migration that fully protects existing investments and minimizes service and
business interruption.
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SIP Trunking
Enterprises can start by taking advantage of new Service Provider IP SIP trunking
service. This is as simple as adding SIP Enablement Services to their existing
Communication Manager (3.0 or above) network. Trunk support conforms to
applicable SIP standards and the SIPConnect specification, an industry initiative to
define a standards-based approach to direct IP peering between SIP-enabled IP
PBXs and VoIP service provider networks. This allows the enterprise to take
advantage of SIP-based PSTN origination/termination services offered at very
competitive prices, without making any other changes to their existing system.
Instant Messaging (IM)/Presence
Once SIP Enablement Services are in place, a gradual step-by-step migration path
allows the enterprise to maximize the benefits of SIP while fully preserving
compatibility and investment protection with existing H.323, digital and analog
endpoints and infrastructure. For example, SIP Enablement Services can be
leveraged to introduce secure Enterprise Instant Messaging (IM) and user presence,
integrated with telephony, through the Avaya IP Softphone and IP Agent client
applications. IP Softphone and IP Agent combine H.323 telephony support with a
SIP-based IM client and a presence-enabled contact directory that supports both
voice and IM communications. This allows enterprises to extend the benefits of user
presence and IM to all users without the need to make extensive changes to their
existing voice infrastructure.
SIP Telephony
SIP supports both numerical (telephony) and alphanumeric addressing, providing a
critical bridge for communications between PSTN and Internet networks. This allows
users on either network to reach any other user without giving up existing devices
or the advantages of each. When enterprises are ready to make the transition to
SIP Telephony, Avaya SIP Enablement Services will allow them fully leverage the
benefits of SIP, while supporting full compatibility and use of their existing
endpoints.
For enterprises that have already deployed H.323 IP Telephony, the migration
process can be initiated immediately and transitioned at whatever pace is desired.
Once registered and licensed on SIP Enablement Services, existing Avaya IP phones
can convert their operation from H.323 to SIP through a simple and free firmware
upgrade. Through Communication Manager Extended Access, SIP endpoints have
access to additional telephony features. IP Softphone users have a similar migration
path to SIP telephony through the Avaya SIP Softphone, which supports SIP for
both IM and telephony.
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A key feature of SIP is its support of Uniform Resource Indicators (URI) for user
addressing, in the same basic form as e-mail addresses (i.e. [email protected]).
Because this addressing is based on the user, not a device, it can be mapped to
whatever device the user desires. Ultimately, this will allow people to communicate
with each other using a single handle-based address vs. the hard-to-remember
multiple phone numbers of their desk phone, cell phone, pager, etc. Through SIP
Enablement Services, communication over even existing telephony networks
become simpler, more intuitive, and focused on the user vs. a device.
User Mobility
SIP is well-suited for mobility requirements. When a user logs onto a SIP device, it
registers the user and sends the SIP URI of the device to the registrar service,
which is used to route calls to/from the user. Avaya Personal Profile Manager, a
user mobility application within SIP Enablement Services, leverages this capability
to create a custom work environment that follows the user. Personal Profile
Manager is centralized service that resides in each SIP Home Proxy Server and
communicates with its assigned SIP endpoints to receive, store, and distribute user
profile information.
SIP Applications
For Avaya, SIP is a catalyst for the evolution of enterprise communications to
Converged Communications, as telephony migrates from a standalone
communication solution to a multi-modal communication core service that can be
integrated with other business applications. Avaya is taking the lead in the
modularization of our software and systems into an open communication
architecture. As solutions become more modular, their services can be deployed in
a greater number of configurations and more easily integrated within a multivendor environment.
Through SIP Enablement Services, a number of Avaya MultiVantage™ Business
Communication Applications, including Communication Manager telephony and
Meeting Exchange web/audio conferencing, become available to a wide range of
standards-based user agents, web-based applications, and communication devices
to create a new paradigm of Converged Communications that will lead to increased
flexibility and cost efficiency.
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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.
Avaya Response:
Linux Servers:
The Avaya Media Servers run a hardened Linux operating system with all
unnecessary/unused services and software removed or disabled. These servers do
not support operator use of a web browser and are thus not subject to infection
from the content of malicious sites. Thus, one of the most popular vectors for virus
transfer is not present.
Alarm reporting and remote maintenance is currently handled through dial-up/dialout access over a modem. Dial-up access to the server requires multiple levels of
strong authentication using a one-time password challenge/response system. In
addition, security precautions are in place to ensure access by Avaya Services is
restricted to only information that is required to carry out a specific role.
Viruses and worms are most commonly delivered via e-mail, by visiting infected
web sites, or by sharing of disk drives. The Avaya S8X00 Media Servers do not
support incoming e-mail, does not support forwarding of E-mail (since there is no
incoming e-mail in the first place), do not contain a WEB browser, and does not
support NFS or SMB (i.e. does not share drives). All file transfers to the Avaya
S8x00 Media Servers are restricted and are cryptographically signed to prevent
introduction of unwanted (trojaned) software.
The Avaya S8X00 Media Server development team investigated (and continues to
investigate as this is not a static area) adding virus scanning software as part of the
software operating on the server. Virus scanning software for Linux is not common,
generally because the market and threats are not sufficiently large to entice
scanner vendors. In addition, those vendors that do offer Linux scanners primarily
scan E-mail (Avaya S8X00 Media Servers have no incoming/SMTP server so there is
nothing for these products to scan) to prevent the spread of a virus to Windows
based machines on the network; they do little specifically for the Linux box itself. A
final and no less important consideration is the impact the introduction of additional
software has on the reliability and overall stability of the platform. The more
software/processes introduced, the greater the potential impact on reliability and
stability. All of these different areas were weighed against the overall risk of
malware being introduced into the system. At this time, the risk of malware
introduction does not merit the inclusion of virus scanning software on the server.
Access:
All access to the S8X00 Communication system requires a valid login on the
communication manager for each user to be able to access.
All products offered through the Integrated Management Suite follow all United
States guidelines for encryption algorithms that can be exported.
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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.
Avaya Response:
Comply. Typical mechanisms of server access include Telnet, Web Browser (HTTP),
and FTP for file transfer. Each of these mechanisms can support login
authentication, but suffer a common weakness. During the login sequence, the
username and password being supplied by the user is sent in clear text. This allows
a person with a network monitor/sniffer to capture the password and gain access.
In addition these mechanisms transmit all the session information in clear text.
Some of this information might contain data such as account codes, authorization
codes or other data useful to an attacker. To overcome these problems, Avaya
S8X00 Media Servers support Secure Shell Access (SSH), Secure Copy (SCP), and
Secure File Transfer Protocol (SFTP) and secure Web access (HTTPS).
¾
SSH, SCP, SFTP provide an access mechanism for terminal access and file
copy that encrypt the entire session including the login sequence as well
as subsequent data transfer.
¾
HTTPS provides a similar mechanism for Web access. All Web access to an
Avaya S8X00 Media Server is via a secure connection. Unencrypted Web
access is not an option.
Security precautions are available on the Linux server to ensure user access is
restricted to only information that is required. Standard Linux operating system
capabilities allow creation of login/password sets and permission groups allowing
only read and/or write access to the partitions of the server that are needed by
each user.
Avaya Communication Manager allows the administrator to modify the permissions
associated with a login. The system maintains default permissions for each level of
login, but the customer may want to further restrict a login, or make sure the
defaults are appropriate for the user. The default values for these fields vary based
on the login type.
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Command Permission Categories Screen
When set to y, the permissions on the Command Permission Categories screen
apply for any object that is not restricted. Additional pages of the Command
Permission Categories screen allow the administrator to restrict the user from any
access to specified objects. If it is necessary to limit a user’s permissions beyond
those on page one, then the objects in this list can be used for further granularity.
Command Permission Categories Screen, page 2 & page 3
For example, if the administrator chooses to allow a user to be able to add and
change stations, but not Vector Directory Numbers (VDNs), they can enter y in the
Administer Stations field and the Additional Restrictions field.
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The system supports two types of administrative users — superuser and nonsuperuser
¾
A superuser provides access to the add, change, display, list, and remove
commands for all customer logins and passwords. The superuser can
administer any mix of superuser/non-superuser logins. The superuser can
administer between 10 and 19 logins depending on the S8X00 system.
¾
Non-superusers may change their password with permission set by the
superuser. However, once a password has been changed, a non-superuser
must wait an interval period before changing their password again (to
prevent password toggling).
¾
Logins must be 3 to 6 alphabetic/numeric characters, or a combination of
both.
¾
The system defines a maximum number of failed login attempts. For instance
the there are 4 login failures in 10 minutes or less, the login will be locked
out for defined interval. An account lockout event is logged to the security
log. This mitigates the threat of brute force password guessing attacks.
¾
The Avaya S8X00 Media Servers continues to support the option of one-timepasswords for logins
On an Avaya S8X00 Media Server, the FTP service is disabled by default. Each time
a file is to be transferred to the server, an administrator must log in and enable the
FTP server (or preferably utilize SFTP instead). The file is then transferred using
anonymous FTP, and the FTP server can then be disabled. Using anonymous FTP in
this manner avoids the problem of sending passwords in clear text.
The server is protected by multiple levels of access to the operating system. For
example, access to a Linux "shell" from which arbitrary commands may be
executed is not granted by default to a login on an Avaya S8X00 Media Server.
When a login is created, the system administrator can specify whether or not the
account is permitted to have shell access. Accounts that are not allowed shell
access receive either an Avaya Communication Manager administration screen or a
Web page upon successful login. In both cases, the operations that may be
performed are restricted. In general, shell access is needed only by individuals that
perform hardware or software maintenance of the server.
On a Linux system the highest level of administrative access is known as "root".
Direct login to a root level account is not permitted on an Avaya S8X00 Media
Server. Administrative access that requires root level permissions is handled via
"proxy" programs that grant specific access to specific accounts to specific
commands and provides detailed logging. The ability to obtain full root level access
is granted only in very special circumstances. This policy helps protect the core
software from security breaches.
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Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
Avaya S8X00 Media Servers support a variety of security monitoring features.
Sessions are automatically disconnected after a period of inactivity. Accounts are
automatically locked out for a period of time as a consequence of consecutive failed
login attempts. Files and directories are monitored and audited by tripwire, a host
based intrusion detection system (HIDS). All login sessions, whether successful or
not, are logged. All user activity is logged. Security events are alarmable events
that are reported in two ways. A maintenance alarm is called out to an Avaya
Maintenance center via an analog modem and/or an SNMP trap is sent to one or
more destinations, as administered.
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-of-Service (DoS) attacks.
Avaya Response:
VoiceCon can isolate the S8X00 servers on their own network that can be physically
independent from VoiceCon’s general network or even logically isolated in its own
VLAN using a non-routable "Private Network", where nobody needs to know their IP
address. This provides additional protection for servers and minimizes the risk of
Denial of Service (DoS) and other threats. Multiple redundant C-LANs can be
connected to multiple different blades on the core Ethernet switches or to different
switches, providing resiliency towards network segment failure and DoS attacks.
The Avaya network interfacing devices (CLANs, Medpro’s, IPSI’s. IP Phones, etc)
have been developed to intelligently discard bogus packets and respond to other
non-core protocols in a measured capacity to prevent CPU starvation DoS attacks.
Additionally, known impossible Ethernet frames/ IP packets are discarded as early
as possible to minimize the CPU resources utilizes. If attacked with a bandwidth
DoS attack, the C-LAN (and other network interfacing devices) will gracefully
recover shortly after the attack has completed. If a C-LAN is unreachable, the
endpoints can automatically look for an alternate one for service. Load sharing is
done automatically between all the C-LANs in a network region.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
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.
Avaya Response:
Comply. The embedded encryption features of the proposed solution provide
encryption of the call control, voice client, and media gateway elements to protect
user privacy. Avaya has openly embraced the National Institute of Standards and
Technology (NIST) FIPS 197 Advanced Encryption Algorithm (AES). The Avaya
implementation of the NIST FIPS 197 Advanced Encryption Algorithm (AES) uses a
128bit encryption key. Avaya utilizes the ITU H.235.5 standard to encrypt key
information elements of the signaling between IP end-points and Avaya
Communication Manager. In addition, TLS is utilized to secure the connections
between Avaya Communication Manager and the media gateways. As it regards
media encryption, the selection of an underlying encryption algorithm has to stand
up to the highly demanding requirements of real time media streams which include
scalability, end-point processing capacity and high tolerance to packet loss and reordering. The encryption-processing overhead must minimize CPU cycles and result
in minimal latency and jitter. In non-technical terms this means that the differences
in voice quality resulting from the encryption/decryption process will be
imperceptible to the end-user. Testing has shown a minimal 1-2% increase in endpoint processor utilization and less than 5ms added end-to-end latency. The
encrypted media remarkably adds no additional IP overhead. Avaya’s pioneering
implementation of Media Encryption offers the following benefits:
1. The ability to support end-to-end media encryption throughout the Avaya IP
telephony endpoint portfolio.
2. The flexibility to determine the encryption algorithm that is used. The Avaya
Labs developed Avaya Encryption Algorithm (AEA) provides 104-bit RC-4 based
encryption to the media stream. The Avaya implementation of the NIST FIPS
197 Advanced Encryption Algorithm (AES) uses a 128bit encryption key.
The architectural implementation of Avaya media encryption allows for the addition
of encryption algorithms and modes (cipher suites) as they are adopted in industry.
The next enhancement will be full support for the recently published Secure RealTime Protocol standard, RFC 3711. Administratively, customers are able to specify
an encryption algorithm, just like they are able to specify different media codecs.
No end-user involvement or adaptation is required to deploy media encryption and
administrative configuration is very straight forward and requires no additional
training.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 - System Performance Requirements
To maintain system interoperability with third party products, Avaya Media
encryption is a configurable option and encryption algorithms can be ordered by
usage preference. Avaya is committed to media privacy, and has taken the lead in
offering media privacy with the necessary framework in place to readily support the
standards based solutions of tomorrow. Avaya Media encryption capabilities have
existed for well over five years, since DEFINITY Call Processing release 10. Media
Encryption has been operating in real world mission critical environments and has
field proven experience.
1.6.4
Physical Interfaces
Vendor Response Requirements
Are there separate physical network interfaces to IPTS administration, control, and voice
transmission signaling functions?
Avaya Response:
Comply. To provide the most secure environment that is possible for the system,
network access should be divided into separate zones of control.
One VLAN can be administered for administrative traffic, one for call
signaling, another for voice bearer traffic, and so on.
¾ Layer 3 boundary devices (routers, layer 3 switches, and firewalls) can be
administered to enforce the corporate security policy on traffic that is
destined for the Avaya S8X00 Media Server, Avaya Media Gateways, or
adjuncts.
¾ Packet filters can permit administrative access only from an administrator’s
PC and to deny access from the Avaya S8X00 Media Server or its gateways
to the corporate LAN while allowing call signaling and bearer traffic from all
IP Telephones appropriate access.
¾
In addition, the Separation of Bearer and Signaling (SBS) feature of Avaya
Communication Manager provides a low cost virtual private network with high voice
quality for VoiceCon, and supports cost avoidance by eliminating some of the
requirements for private leased lines.
The SBS feature supports:
¾ QSIG private networking signaling over a low cost IP network.
¾ Voice (bearer) calls over the public switched network.
¾ Association between QSIG feature signaling information and each voice call.
¾ The S8720 Media Server has five interfaces the enterprise LAN and control
LANs are connected together
¾ There is only one control LAN
¾ There are two spare NICs that are not used
¾ The messages between the Media Server and the Media Gateways are
encrypted
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Part 1 - System Performance Requirements
Recommendations for network security
Avaya recommends that these servers be located behind a firewall. Where this
firewall is located with respect to other LAN components must be designed on a
case-by-case basis. Avaya Professional Services can assist customers in configuring
their networks for both security and optimal IP Telephony operation.
Other vendors also specialize in this type of consulting. Owners are advised to seek
assistance if internal staff is not trained in these areas. Security oversights that
arise from negligence, ignorance, or the pressures of schedule or budget are all
equally usable by hackers. Malicious activity is a moving target, and what is safe
today might not be safe tomorrow. Avaya is committed to providing appropriate
secure solutions for its products, and to continuously monitoring evolving security
threats. Avaya S8X00 servers are appropriately secure against the known threats
and hardened to be resilient against threats to come. Avaya responds quickly
should new threats appear and proactively notifying customers via security bulletins
which classify the vulnerability and offer expected remediation time frames.
Consult these resources for the latest security information:
¾
Your Avaya account team
¾
The Avaya Security support Web site: http://support.avaya.com/security.
1.6.5 Root Access
Vendor Response Requirements
Is there direct Root access to the IPTS common control?
Avaya Response:
On a Linux system the highest level of administrative access is known as "root".
Direct login to a root level account is not permitted on an Avaya S8X00 Media
Server. Administrative access that requires root level permissions is handled via
"proxy" programs that grant specific access to specific accounts for specific
programs with extensive logging of such commands. The ability to obtain full root
level access is granted only in very special circumstances. This policy helps protect
the integrity of core software.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 2 – IPTS Network Port Capacity Requirements
2.0
IPTS Network Port Capacity Requirements
The proposed IPTS must be capable of supporting port capacity requirements for the
HQ facility and remote branches. It must also be capable of supporting future VoiceCon
growth requirements at HQ and RO facilities.
Avaya Response:
Comply. The recommended system meets and/or exceeds all requirements.
2.1.0 Port Capacity Requirements
The equipped port capacity of the proposed VoiceCon HQ IP Telephony System at time
of installation and cutover must support of a mix of IP telephones, analog telephones,
facsimile terminals, modems, 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 describes the port capacity requirements for each of the VoiceCon network
locations. Satisfying these stated port capacity requirements is a MANDATORY
requirement
Avaya Response:
Comply. The recommended system meets and/or exceeds all requirements.
2.1.1 HQ Facility
Avaya Response:
Comply. The recommended system meets and/or exceeds all requirements.
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Request for Proposal for an IP Telephony System
Part 1 – Section 2 – IPTS Network Port Capacity Requirements
2.1.2 RO Facility
The RO facility will be a two floor facility that will support:
* 205 desktop 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.
Avaya Response:
Comply. The recommended system meets and/or exceeds all requirements.
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 audio conferencing units;
* 5 analog telephones including 2 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.
Avaya Response:
Comply. The recommended system meets and/or exceeds all requirements.
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Request for Proposal for an IP Telephony System
Part 1 – Section 2 – IPTS Network Port Capacity Requirements
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,
* 4 IP audio conferencing 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.
Avaya Response:
Comply. The recommended system meets and/or exceeds all requirements.
2.1.5 SB3 (Small)
The SB3 facility will be a single floor facility that will support:
* 8 Desktop 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
Avaya Response:
Comply. The recommended system meets and/or exceeds all requirements.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 2 – IPTS Network Port Capacity Requirements
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
Avaya Response:
Comply. The recommended system meets and/or exceeds all requirements.
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 – Section 2 – IPTS Network Port Capacity Requirements
2.3
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
HQ
RO
SB1
SB2
SB3
T-1 Digital Local
Inbound/Outbound
6
2
1
0
0
T-1 Digital
Long
Distance
7
2
0
0
0
Analog
(PFTS)
5
2
2
2
1
2-way
GS/LS
25
10
5
10
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.
Avaya Response:
Comply with explanation. Avaya has included the requested capacity and growth
potential for the HQ, RO, SB1 and SB2 sites. In analyzing the trunk request for the
SB3 branch Avaya found VoiceCon is requesting a 50% trunk to users’ ratio while
the other sites average 33% trunks to users.
Based on this review Avaya is respectfully proposing our G250 Media Gateway for
the SB3 location. The G250 has a four trunk capacity, not five as requested. This
will still give VoiceCon a 40% ratio of stations to trunks locally as well as access to
other locations in the network based on the ARS designed at time of
implementation. Avaya feels this gateway will meet and exceed the SB3 site needs.
The proposed solution is designed to support centralized long distance trunk
resources for branch locations, using our World Class routing features, the use of
HQ Trunking by the branches is completely customizable.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 2 – IPTS Network Port Capacity Requirements
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.
Avaya Response:
Comply. The recommended system meets and/or exceeds all requirements.
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 facilities 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.
Avaya Response:
Comply. The recommended system meets and/or exceeds all requirements.
System Ports
Port Capacity of Proposed IPTS
Total Stations
36,000
IP Stations
12,000
Analog Stations
36,0001
Total Trunk Circuits
8,000
Analog Circuits
8,0002
T1-carrier Interfaces
4003
Trunk Groups
2000
Trunks Circuits/Group
1
Any portion of the maximum number of stations can be analog
2
Any portion of the maximum number of trunks can be analog
3
255
An additional 166 DS1 interfaces are permitted (for a total of 566), which can be used only for Line Side DS1 connections, not as
trunks.
December 1, 2006
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VoiceCon
Request for Proposal for an IP Telephony System
Part 1 – Section 3 – Port Interface & Handling Requirements
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; audio
conferencing 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.
Avaya Response:
Comply. The
requirements.
3.1
proposed
solution
is
engineered
to
meet
or
exceed
these
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.
Avaya Response:
Comply. The proposed solution will support analog, IP, SIP as designed. Traditional
digital voice terminal equipment can be added via a circuit pack addition
The Avaya S8720 Media Server can support up to a system maximum of 15,424
simultaneous circuit switched calls or 242 per port network. There are 484 time
slots for voice and data per port network. For IP to IP calls this limitation does not
apply. The CCS rating for the S8720 Media Server is 600,000 general business type
Busy Hour Call Completions.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 3 – Port Interface & Handling Requirements
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.
Avaya Response:
Comply. Direct IP to IP audio permits calls between IP endpoints to “shuffle”
directly to each other, instead of speaking through the MedPro board or VoIP
module. If a feature that requires the media gateway, such as conferencing, is
activated during the call, the endpoints shuffle back to the MedPro board or VoIP
module. At the conclusion of the conference when only two IP endpoints remain,
the IP stations shuffle back to one another.
When direct IP to IP audio is enabled, it applies to both intra-region and interregion calls, however, an additional step is required to permit inter-region shuffling
– an inter-region codec set must be pre-selected. When a call is made between the
inter-connected regions, the specified codec set is used.
As an example, the IP telephones in network region 1 use codec set 1 because they
are in a LAN environment. The IP telephones in network region 2 use codec set 2
because they must traverse the WAN to access the G650 media gateway resources.
Inter-region (1-to-2) calls use codec set 2 because they take place over the WAN.
In a VoIP solution, sometimes it is necessary to provide signaling between different
CODECs. In this case, transcoding of the CODEC is handled by using the resources
of the gateways to do the transcoding. An example is that if an endpoint can only
speak G.711 and another endpoint can only speak G.729, then the two endpoints
are terminated at the gateways in the configuration, transcoded, and connected.
Obviously, if two endpoints have the ability to adjust their CODEC selection and can
speak to the same CODEC, then the endpoints negotiate CODEC selection and
speak directly over the data network without utilizing the gateway functionality.
Sometimes IP endpoints do not support compatible audio CODEC or an endpoint
does not support the shuffling message sequence in H.323, in which case the Avaya
Server is forced to converge audio on a Media Processor in order to complete an
audio connection. When this activity involves two endpoints in the same region, the
Avaya Server will use a technique called "hairpinning" to set up the IP voice path.
"Hairpinning" is a technique to converge voice through the Media Processor without
consuming gateway TDM bus resources.
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Request for Proposal for an IP Telephony System
Part 1 – Section 3 – Port Interface & Handling Requirements
For example, suppose two endpoints (E1 and E2) in the same region are involved in
a call and E1 supports only G.711 and E2 supports only G.729. The Avaya Media
Server will direct both endpoints to connect their audio to a selected media
processor. The media processor will terminate the audio stream from E1; pass it
through jitter buffers and G.711audio decoders as described earlier. The media
stream is then passed into the G.729 encoder and transmitted to E2. This
"hairpinning" of the audio stream automatically inserts DSP resources and keeps
the call in an IP format. This technique is called "deep hairpinning".
Other audio configurations allow the Avaya Media Server to save additional media
processor resources. Suppose two endpoints in the same region are involved in a
call, and both support G.711, but one endpoint is NetMeeting. Generally, the Avaya
Server would use shuffling to establish the audio connection between these
endpoints. However, NetMeeting does not support the message sequence for
shuffling, so the Avaya Server instead directs both endpoints to send their audio
streams to a media processor which "hairpins" the audio streams out at the NIC
level. This process sends all received audio packets to the other side without
sending them through jitter buffers or decoders. This technique is called "shallow
hairpinning" and saves DSP and buffer resources on the media processor.
3.2.1 IP Station Discovery
How do IP communications devices learn about their voice VLAN, including IP
addresses, default gateways, call controller, TFTP server, QoS settings, VLANs, and
other parameters. Does the proposed system solution employ proprietary protocols for
IP communications devices to learn their voice VLAN or is an industry standard, such as
Dynamic Host Control Protocol (DHCP) used?
Avaya Response:
Avaya IP Telephones can use Dynamic Host Configuration Protocol (DHCP) standard
to learn their IP addresses, default gateways, call controller, TFTP server, QoS
settings, VLANs, and other parameters. Avaya can use a special option, Option 176,
to pass these values. Avaya has done significant testing of and had good success
with Option 176 on the Microsoft Windows 2000 DHCP server and the ISC DHCP
server (common on Linux and UNIX platforms). Results from other DHCP servers
may vary. A typical Option 176 string looks like the following string:
“MCIPADD=#.#.#.#,MCPORT=1719,TFTPSRVR=#.#.#.#,L2Q=1,L2QVLAN=0”
where
¾
MCIPADD is the IP address of the C-LAN
¾
MCPORT is the UDP port that is used for telephone registration
¾
TFTPSRVR is the TFTP server that the telephone uses to look for firmware
and configuration upgrades
¾
L2Q is 802.1Q. 1 is on, 0 is off
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Request for Proposal for an IP Telephony System
Part 1 – Section 3 – Port Interface & Handling Requirements
¾
L2QVLAN is the VLAN that the telephone uses. VLAN ID 0 is a special VLAN
ID that tells the next Layer 2 switch to replace the 0 tag with the native
VLAN ID on that ingress port.
A gatekeeper is an H.323 entity on the network that provides address translation
and controls access to the network for H.323 endpoints. For Communication
Manager systems, these are the Avaya Media Servers through the C-LAN circuit
packs. H.323 RAS (Registration, Admission, and Status) Protocol messages are
exchanged between them and the IP endpoints for the endpoint registration.
All IP endpoints (IP SoftPhones, IP agents, and IP Telephones) H.323 voice
applications register with an Avaya gatekeeper before any calls are attempted.
Communication Manager enforces call signaling (Q.931) and call control (H.245)
channels from endpoints to terminate on the gatekeeper. This allows
Communication Manager to provide many of its calling features to H.323 calls.
The RAS protocol is used by the IP endpoint to discover and register with the
Communication Manager gatekeeper. The discovery mechanism uses unicast IP
facilities. When registration with the original gatekeeper (C-LAN or S8300) IP
address is successful, the switch sends back the IP addresses of all the C-LANs (or
LSPs) in the IP Telephone’s network region. These addresses are used if the call
signaling to the original C-LAN circuit pack fails.
Discovery and Registration Process to the Gatekeeper
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The Avaya 9600/4600 Series IP Telephone looks and feels just like a circuitswitched phone because it supports most of the features of Avaya digital
telephones. This family of phones supports IEEE802.1Q/p, DiffServ and a separate
VLAN for voice traffic. It can withstand over 1,000 broadcasts per second making it
an excellent phone in resisting broadcast storms. It also has a full-duplex
speakerphone. It also supports traffic at 10 or 100 Mbps, supports silence
suppression, and can use DHCP (Dynamic Host Configuration Protocol) for easy
setup. If auto-negotiation is not used, the switched port should be 10 or 100 Mbps
and half-duplex.
Avaya IP Telephones operate with industry-standard protocols and advanced
software, including the H.323 and SIP protocols for voice communication over an
Internet Protocol network, G.711 and G.729A/B voice coders and a variety of
Quality of Service capabilities. Those include UDP Port selection at Layer 4, DiffServ
priority numbering at Layer 3, and 802.1p/q at Layer 2. Avaya IP telephones also
support the 802.3af power over Ethernet standard.
IP Telephones and media gateways receive a list of alternate gatekeepers as part of
the standard configuration. If the endpoints cannot reach their primary gatekeeper,
the endpoints will re-register with the secondary gatekeeper. If they cannot reach
that gatekeeper, they will attempt to register with the tertiary gatekeeper, and so
on.
Avaya provides multiple signaling channels by the deployment of C-LAN cards. This
is one of the most important competitive advantages in our architecture. End-points
and remote Media Gateways (G700) register with one of the multiple C-LAN cards
in order to get telephony services. Moves, Adds and Changes are done that way as
well. Because the endpoints and computers talk to one of those multiple signaling
channels instead of talking directly to the servers, customers can isolate the S8720
Media Servers in their own network that can be physically independent from the
customer's network or even logically isolated in its own VLAN using a non-routable
"Secret Network", where nobody needs to know their IP address. This provides
awesome protection for servers, dispensing the need for virus updates and pretty
much eliminating the risk of Denial of Service (DoS) and hacker attacks. By having
multiple redundant C-LANs, we can connect them to multiple different blades on the
core Ethernet switches or to different switches, providing resiliency towards
network segment failure and DoS attacks. If attacked, the C-LAN recovers shortly
after the attack is done. If you lose a C-LAN, the endpoints can automatically look
for an alternate one for service. Load sharing is done automatically between all the
C-LANs in a network region.
The IP telephones support IEEE 802.1X as a supplicant using MD5 and the EAP
authentication. Proxy EAPoL log-off is configurable (to automatically log-off the PC
plugged into the phone when link-loss is detected).
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The H.323 phones and H.323 IP Softphone support Avaya Media Encryption today,
which is based on the SRTP standard (RFC 3711) and full compliance. Media
Encryption offers the option of the 104 bit RC4 based Avaya Encryption Algorithm
(AEA) or the 128bit AES Encryption Algorithm in Counter Mode Encryption (CTR)
mode. Avaya Media Encryption has been shipping for over four years and provides
protection of the Real Time Protocol (RTP) payload in a wide variety of
configurations (peer-to-peer, peer-to-media gateway, via IP Softphone, across
independent Media Servers, etc). Note that other vendors run into significant
scalability problems (capacity limitations) when deploying media encryption on a
large scale and also limit media encryption to point to point calls (no 3 or more
party calls). In addition, specific models of other vendors’ phones and gateways and
do not support media encryption on Softphones
The H.323 phones use ITU Standard, H.235.5 for signaling encryption which
provides confidentiality of the display data, DTMF digits, and authentication
credentials in addition to packet integrity. H.235.5 uses the Personal Identification
Number (PIN), a pre-shared secret between the user and the Media Server to
authenticate both sides of the conversation. The PIN code can be defined in Avaya
Communication Manager with a size of 3 to 8 decimal digits.
The IP also support a challenge/response authentication process (a less secure
alternative to the newer H.235.5 option - both options are configurable by the user)
which has the IP phone hash a challenge plus its PIN and send it to the Media
Server, which performs the same operation to ensure the user has knowledge the
pre-shared secret without sending the secret across the network.
Avaya IP Phones support 802.1ab, (also referred to as Link Layer Discovery
Protocol). LLDP, which is the IEEE standardized protocol designed to replace
significantly less secure and proprietary L2 discovery protocols currently in the
marketplace, LLDP also provides VoiceCon more choices when it comes to selecting
a vendor.
IP Sets have a READ ONLY MIB (SNMPv2c) - SNMP sets have no impact on the
phone’s behavior. In addition, to provide continuity across the enterprise and
prevent user-caused quality issues, access to the SNMP agent in the phone can be
restricted to a particular list of IP addresses (for SNMP management stations) and
the READ ONLY community string is customer configurable.
End-user access to the configuration information of the phone (i.e. IP Address, Call
Server, etc) can be administratively restricted - so the user cannot view these
variables without a required password.
Avaya H.323 and SIP phones provide the capability to download their configuration
files via HTTPs, validating the HTTPs server using public key cryptography.
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For remote and work at home employees, Avaya offers an integrated VPN client
which runs in the IP phone. This solution encapsulates traffic inside an IPSec
tunnel. A home office-based worker can simply plug the phone into a power source,
connect it to a home broadband router, enter a password, and it is operable. No
other equipment or software is required in the home office. This simplicity can also
be applied to temporary remote deployments during emergencies.
Although not a security related standard from the IEEE, Avaya IP phones fully
support the separation of voice traffic from data traffic using the IEEE 802.1Q
standard.
Additionally, Avaya IP phones do not rely on a web services running in the phone to
provide performance statistics, capability to log in and out, and user phone
administration, since this can open up significant security holes in the network.
While some vendors discuss use of these capabilities, their security blueprint
recommends disabling the web services to secure the solution.
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 inline of IP telephone equipment. Describe current, future and retrospective compatibility
of all proposed equipment. If 802.3af is not supported, identify the PoE implementation
being proposed.
Avaya Response:
Avaya, as an industry leader and visionary, has actively participated in the
development of the IEEE 802.3af standard since its inception several years ago. In
June of 2003 the IEEE approved 802.3af now known as "802.3af-2003." All IP
telephones currently being sold by Avaya are compatible with the standard. We
remain committed to designing our products to be compliant with the specifications
of the standard as is demonstrated by our leadership position within the committee
responsible for the recommendation.
Our strength lies in the number of different options that we can provide compared
to our competitors. These include:
¾ Avaya 4600 and 9600 Series IP Telephones that are designed to comply with
the specifications of the IEEE 802.3af standard
¾ Avaya Mid-Span Power Solutions (a patch panel type device that adds
802.3af power capability to non-802.3af compatible Ethernet switches available in 1, 6, 12 and 24 ports)
¾ Avaya P333T-PWR layer 2 switch designed to support 802.3af power
¾ Power over Ethernet (PoE) blade for the C460 multi-layer chassis based
switch designed to support 802.3af power
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¾ The 1151B1 individual external power supply and the 1151B2 which has built
in battery backup for IP phones
¾ An adapter which allows customers to power newer Avaya telephones from
certain Cisco switches that only provide proprietary power
¾ The Avaya AP3 (2nd generation), AP4, AP5, and AP6 Access Points that can
be powered by an IEEE 802.3af compatible device
Avaya IP telephones have been and continue to be compliant since May of 2002.
Avaya is committed to developing standards based solutions that allow the
customer the flexibility to choose solutions based on their business needs as
opposed to forcing them to one manufacturer.
3.2.3 IP Station QoS
Vendor Response Requirement:
Describe the proposed IPTS solution’s capabilities to provide Layer 2 and Layer 3 QoS
to IP stations to ensuring end-to-end quality of service. Include in the response what
industry standards are deployed.
Avaya Response:
QoS for voice packets is obtained only after a Class of Service (CoS) mechanism
tags voice packets as having priority over data packets. IEEE 802.1p/Q at the
Ethernet layer (Layer 2) and DiffServ Code Points (DSCP) at the IP layer (Layer 3)
are two standards based CoS mechanisms that are used by Avaya products. These
mechanisms are supported by the IP Telephone, the S8300 Media Server, and the
C-LAN and MedPro circuit packs. Although TCP/UDP source and destination ports
are not CoS mechanisms, they can be used to identify specific traffic, and can be
used much like CoS tags. Other non-CoS methods to identify specific traffic are to
key in on source and destination IP addresses and specific protocols, such as RTP.
The MedPro circuit pack and IP Telephones use RTP to encapsulate audio.
Class of Service refers to mechanisms that tags traffic in such a way that the traffic
can be differentiated and segregated into various classes. Quality of Service refers
to what the network does to the tagged traffic to give higher priority to specific
classes. If an endpoint tags its traffic with Layer 2 802.1p priority 6 and Layer 3
Differentiated Services Code Point (DSCP) 46, for example, the Ethernet switch
must be configured to give priority to value 6, and the router must be configured to
give priority to DSCP 46. The fact that certain traffic is tagged with the intent to
give it higher priority does not necessarily mean it will receive higher priority. CoS
tagging is not effective without the supporting QoS mechanisms in the network
devices.
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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 multiparty add-on 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.
Avaya Response:
Avaya Communication Manager natively supports conferencing capabilities with up
to five other conferees without adding additional hardware or software (no
additional servers are required.) Users with multi-line telephones can set up a
conference call between conferees either internal or external to the system, and
can be local or long distance.
Users can toggle or swap connections to multiple conference parties (alternately
placing each called party on soft hold) with the Toggle/Swap button. The caller can
still press the Conference button to conference with all the called parties, or can
press Transfer to drop his/her own connection, thereby conferencing only the
others (called parties). The Selective conference party display and drop (or forced
release on the attendant console) feature uses repeated presses of the Conference
Display button to cycle through the display of the names and numbers (if available)
of all parties on the call. The caller may selectively drop or mute any party on the
conference.
The solution also natively supports “meet-me” conferencing, via programming a
special Vector Directory Number (VDN), secured via an access code, and to allow
up to six parties to join a conference. No additional hardware, software, or
additional circuits are required.
With IP endpoints, the conferencing feature requires interaction with the media
gateway, so if this feature is activated during a call between two IP endpoints, they
shuffle back to the MedPro board or VoIP module., at the conclusion of the and only
the two IP parties remain, the IP stations shuffle back to a direct IP to IP
connection.
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3.4
VoIP Overflow Traffic
If available WAN circuits connecting the HQ, RO and all SB facilities are busy, call
admission control levels are reached, or QoS levels are not satisfied on-network voice
traffic must be able to automatically overflow to PSTN trunk circuits.
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 cab revert back to the WAN if conditions
permit.
Avaya Response:
Comply. Avaya Communication Manger supports the bypass of IP trunks if QOS
levels are not met. The specific metrics that can be set are Round Trip Time and
packet loss. An administrator can set thresholds for the bypass operation and can
also set different thresholds for the re-establishment of the normally used trunks.
In the case of trunk circuits being busy an administrator can choose to re-route
over alternate trunking facilities by specifying a secondary route within the route
plan form. All of the bypass functionality works independently of the type of trunk
used for bypass (i.e. analog, ISDN).
3.5.0 Port Interface Circuit Cards
For each of the following port types, provide a brief description of the proposed port
interface circuit card(s) and/or media gateway equipment included with the proposed
IPTS to support analog, digital, and IP ports. Include in the descriptions below the
number of port interface terminations for each port circuit card, and the number of
available gateway channels for each media gateway unit.
Avaya Response:
Comply. Responses included as part of the items described in Sections 3.5.1 thru
3.5.8.
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.
Avaya Response:
Comply. Methods of supporting IP telephones (based on the Media Gateway) are
provided below.
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Avaya G650 Media Gateway
The TN2302AP IP Media Processor Circuit Pack provides VoIP audio access to the
system for local stations and for outside trunks. The TN2302AP can perform echo
cancellation, silence suppression, FAX relay service, and DTMF detection. The
TN2302AP is the H.323 audio platform, includes a 10/100 BaseT Ethernet interface,
supports the T.30 and T.38 standards for FAX transmission, and is firmware
downloadable.
The TN2302AP provides audio processing for between 32 and 64 voice channels,
depending on the CODECs in use. The TN2302AP supports hairpin connections and
shuffling of calls between TDM connections and IP-to-IP direct connections.
Avaya G700 Media Gateway
The Avaya G700 Media Gateway uses a VoIP Motherboard with the following
capabilities:
¾
8 channels of tone detection
¾
Tone generation
¾
Clock generation/synchronization
¾
20 minutes of announcement storage
¾
15 announcement playback channels
¾
LAPD termination
¾
H.248 signaling to the Controller
¾
Gateway maintenance
I960 CAJUN Complex
¾
Based on P330 CAJUN switch
o
Eight-port Layer 2 switch ASIC
o
PCI Bus
o
Cascade Module slot – up to ten Media Gateways or P330s can be
stacked
¾
Expansion Module slot (not hot-pluggable)
¾
802.1p/q VLAN tagging
The Avaya VoIP MM760 Media Module is a clone of the motherboard VoIP engine.
The VoIP MM760 provides an additional 64 VoIP channels with G.711 compression.
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The capacity of the VoIP MM760 is 64 G.711 TDM/IP simultaneous calls, or 32
compression codec, G.729 or G.723, TDM/IP simultaneous calls. These call types
can be mixed on the same resource. In other words, the simultaneous call capacity
of the resource is 64 G.711 Equivalent Calls.
3.5.2 Analog telephones
Vendor Response Requirement:
Provide a brief description how analog telephones are logically and physically supported
by the common control call telephony server, identifying all intermediary hardware
elements necessary for control signaling transmission. Specify the number of circuit
terminations per circuit board/module/media gateway.
Avaya Response:
Comply. The methods of supporting analog telephones (based on the Media
Gateway) are described below.
G650 Media Gateway – Analog Stations
Analog stations are supported at the HQ site using the TN793B circuit pack, a dual
coded, analog line 24-port circuit pack that supports Caller ID telephones and Caller
ID devices. Each port supports one telephone, such as 500 (rotary dial) and 2500
telephones (DTMF dial).
The TN793B supports on-premises (in-building) wiring with either touch-tone or
rotary dialing and with or without the LED and neon Message Waiting Indicators.
The TN793B circuit pack supports off-premises wiring with either DTMF or rotary
dialing, but LED or neon message waiting indicators are not supported off-premises.
The TN793B circuit pack, along with a TN755B neon power circuit pack supports onpremise telephones that are equipped with neon message waiting indicators. The
TN793B supports three ringer loads, only one telephone can have an LED or neon
message waiting indicator. The TN793B circuit pack allows a maximum of 12
simultaneous ports ringing; four on ports 1 through 8, four on ports 9 through 16,
and four on ports 17 through 24.
The TN793B circuit pack supports A-law and µ-law companding and administrable
timers. The TN793 circuit pack supports queue warning level lights associated with
the DDC and the UCD features, recorded announcements associated with the
Intercept Treatment feature, and a paging system for the Loudspeaker Paging
feature. Additional support is provided for external alerting devices that are
associated with the Trunk Answer Any Station (TAAS) feature, neon message
waiting indicators, and modems The TN793B also supports secondary lightning
protection. The TN793B provides -48 V DC current in the off-hook state. Ringing
voltage is -90 VDC.
The following table lists the TN793B-supported telephones and shows each of their
wiring sizes and ranges.
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Telephone
Wire size (AWG)
Maximum range
(feet)
500-type
24 (0.2 mm2/0.5 mm)
20,000 (6,096 m)
2500-type
24 (0.2 mm2/0.5 mm)
20,000 (6,096 m)
6200-type
24 (0.2 mm2/0.5 mm)
12,000 (3,657 m)
7100-series
24 (0.2 mm2/0.5 mm)
20,000 (6,096 m)
8100-series
24 (0.2 mm2/0.5 mm)
12,000 (3,657 m)
9100-series
24 (0.2 mm2/0.5 mm)
12,000 (3,657 m)
G700 Media Gateways – Analog Stations
Analog stations are supported using the Avaya MM711 Analog Media Module, which
provides eight analog telephone or trunk ports. See the following illustration for an
example of the MM711.
Avaya MM711 Analog Media Module
The MM711 provides the administrator with the capability to configure any of the
eight ports of this analog circuit pack as:
¾
A loop start or a ground start central office trunk
o
Loop current 18-60mA
¾
A wink start or a immediate start Analog Direct Inward Dialing (DID) trunk
¾
A 2-wire analog Outgoing CAMA E911 trunk, for connectivity to the PSTN
o
¾
MF signaling is supported for CAMA ports
Analog, tip/ring devices such as single-line telephones with or without LED
message waiting indication
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The MM711 also supports:
¾
Type 1 and Type 2 Caller ID
¾
Ring voltage generation for a variety of international frequencies and
cadences
o
ALM
o
TST
o
ACT
o
MM716
o
ANALOG
o
VH0
MM716 Analog Media Module
The MM716 provides 24 analog ports supporting telephones, trunks, modem, and
fax. The 24 ports are provided via a 25 pair RJ21X amphenol connector, which can
be connected by an amphenol cable to a breakout box or punch down block. These
ports can be configured as DID trunks with either wink start or immediate start.
See the following figure for an example of the MM716.
The MM716 provides you with the capability to configure any of the 24 ports as:
¾ A wink-start or an immediate-start DID trunk
¾ Analog tip/ring devices such as single-line telephones with or without LED
message waiting indication
The MM716 also supports:
¾ Three ringer loads, which is the ringer equivalency number for up to 2,000
feet (610 meters) for all 24 ports
¾ Up to 24 simultaneously-ringing ports
¾ Type 1 caller ID and Type 2 caller ID
¾ Ring voltage generation for a variety of international frequencies and
cadences
The MM716 is compatible with ACM version 2.0 and higher and Communication
Manager Firmware version 25.0.0 and higher.
G250 Media Gateway – Analog Stations
G250 supports two analog ports that are inherent in the chassis.
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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.
Avaya Response:
Comply. T.38 has emerged as the key standard for ensuring that VoIP capable
networks can continue to support user requirements for real-time fax
communications, and enables enterprises to support fax transmissions over IP in a
multi-vendor environment. T.38 Fax relay is the ITU-T defined standard for Fax
relay. It involves H.323/H.245 signaling capabilities exchange for Fax relay. Fax
bearer traffic is relayed (encoded/decoded) according to the T.38 specification.
Avaya has incorporated this standard into its Media Processor TN2302BP circuit
packs and VoIP media modules for media gateways so that they will interoperate
with non-Avaya systems such as the Clarent or Alcatel Gatekeeper with Cisco and
Clarent Gateways. Avaya Communication Manager systems can also use T.38 to
transmit Faxes between other Avaya Communication Manager systems and Modular
Messaging. Selection of T.38 for fax transmission involves Avaya Communication
Manager negotiating with the far end system to ensure it will support T.38.
Avaya Media Gateways communicate with the fax devices using the standard ITU-T
T.30 protocol used by all analog fax devices today. Using modem modulation, the
sending fax machine sends T.30 protocol and fax image data to the media gateway,
which demodulates the signals and repackages them into T.38 packets. The T.38
protocol provides an ITU-standard mechanism for a gateway/controller to inform
another gateway of the desire to change the media stream from a voice stream to a
data stream. Communication Manager provides this signaling to external T.38enabled gateways to negotiate the session. The sending gateway then sends the
T.38 packets to the receiving T.38-enabled gateway over IP (UDP), which then
delivers the packets using the T.30 protocol to the endpoint fax device.
The Avaya implementation of the ITU’s T.38 standard is intended to interoperate
with any T.38 compliant product. Remember this means relay mode only. There is a
difference between compliance to a specification and certification from Avaya.
Avaya has tested interoperability with Cisco version 2 gateways, Clarent
gatekeeper, Alcatel gatekeeper, and has certified Multi-Tech.
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General Fax/Modem Call Flow for Avaya Communication Manager 2.1.1 and
later
1. The originating analog device (Fax/Modem/TTY/STU) starts a call to a
destination.
2. Communication Manager routes the call to the called number and establishes a
call session as an audio call.
3. Once the audio call is established, fax/modem tones can be exchanged. These
tones indicate whether the call is a Fax, Modem or TTY.
4. If the call is T.38 and is administered, the encoding of the call is changed to
T.38 mode, and Communication Manager signals to the gateways to use T.38
procedures to negotiate a switch-over to T.38 fax mode. If the gateway at the
opposite end of the call acknowledges the T.38 mode request, the initial audio
channel is closed and a T.38 Fax Relay channel is opened. When the fax
transmission is completed, the call is reverted back to voice mode. Similar
procedures are followed for Avaya proprietary fax/modem relay, which is the
default mode between Avaya media gateways.
5. If the call is administered as Pass-Through, the media processor simulates a
“clear channel” by switching the existing voice coder on the fly to G.711 data
mode to increase bandwidth, removes VAD, echo cancellers and set buffers to
reduce jitter.
6. At conclusion of call, resources are released for use by other calls.
General Fax/Modem Call Flow for Avaya Communication Manager
Modems and faxes can also be supported as analog endpoints.
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3.5.4. Modem
Vendor Response Requirement:
Provide a brief description how modem terminals are logically and physically supported
by the common control call telephony server, identifying all intermediary hardware
elements necessary for control signaling transmission. Specify the number of circuit
terminations per circuit board/module/media gateway.
Avaya Response:
Comply. Please refer to item 3.5.3.
3.5.5 Power Failure Transfer Station (PFTS)
Vendor Response Requirement:
Provide a brief description how analog telephone instrument Power Failure Transfer
Stations (PFTSs) are logically and physically supported by the common control call
telephony server, identifying all intermediary hardware elements necessary for control
signaling transmission. Specify the number of circuit terminations per circuit
board/module/media gateway.
Avaya Response:
Comply. Power failure transfer station is handled differently dependent on the
Media Gateway. The Avaya G650 Media Gateway uses traditional PFT units in the
wall field to support PFTS locations to directly connect a designated analog station
and trunk on the system. The PFT units are included in the proposed design. The
TN2312BP IPSI provides environmental maintenance for the G650, including
emergency transfer control. Emergency transfer control provides -48VDC to operate
an external emergency transfer panel. Avaya Communication Manager controls the
state of the emergency transfer. (Note that, in the past, hardware boards or alarm
panels provided a 3-position physical switch to control emergency transfer.)
Avaya Media Gateways use Emergency Transfer (ETR) logic to support power failure
transfer, as described below.
¾
The Emergency Transfer logic, can
connect the chassis ‘trunk’ port to
one of the line ports
¾
When ETR is on, the chassis trunk
port 1 is connected to line port 1
and line port 2 is disconnected
¾
When ETR is on, the chassis analog
ports are unmanaged by the call
controller
¾
The Media Gateway reports to Avaya CM about V7 presence only when ETR is
off
¾
ETR will be ON when Power to the Media Gateway is OFF
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¾
Inability to contact any of the call controllers
¾
ETR is administratively ON (CLI, SNMP) Severe software fault
¾
ETR will be OFF when ETR is administratively OFF
¾
ETR is administratively AUTO and there is a connection between the Servers
and an ACM (providing the Current Loop detector shows there are no ongoing
calls)
¾
After reboot, the Media Gateway will wait until the Current loop Detector
shows that there are no active calls between the trunk and the line before
reporting ‘V7 insertion’ to Avaya CM
¾
ETR Special LED Indication
For emergency power failure transfer at all G700 locations VoiceCon will need to
supply a 48VDC 90 milliamp power supply.
Standard Local Survivability using an Avaya G250 Media Gateway
For very small branch locations, (under 12 users) the Avaya G250 Media Gateway
supports Standard Local Survivability (SLS). SLS is a configurable software module
that allows a local G250 to provide a limited number of Communication Manager
features when no link is available to the server, an LSP, or an Enterprise Survivable
Server (ESS). SLS can be configured on a system-wide basis using the Provisioning
and Installation Manager (PIM) component of Avaya Integrated Management, or
SLS can be configured on an individual G250 using the command line interface
(CLI).
3.5.6 GS/LS CO Trunk
Vendor Response Requirement:
Provide a brief description how GS/LS CO trunk circuits are logically and physically
supported by the common control call telephony server, identifying all intermediary
hardware elements necessary for control signaling transmission. Specify the number of
circuit terminations per circuit board/module/media gateway.
Avaya Response:
Avaya G650 Media Gateway
The TN747B CO trunk circuit pack has eight ports for loop-or ground-start CO,
foreign exchange (FX), and wide area telecommunications service (WATS) trunks.
Each port has tip and ring signal leads. A port can connect to a PagePac paging
system. The TN747B supports the abandoned call search feature in automatic call
distribution (ACD) applications (if the CO has this feature). Vintage 12 or greater of
the TN747B circuit pack also provides battery-reversed signaling.
Avaya G700 Media Gateways
CO trunk circuits are supported using the Avaya MM711 Media Module, which
provides analog trunk and telephone features and functionality. Please refer to item
3.5.2 for a complete description.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 3 – Port Interface & Handling Requirements
Avaya G250 Media Gateways
Is a converged All-in-One Unit: The G250 integrates telephony, routing, and data
switching into one box, reducing the overhead and complexity of managing
disparate gateways, routers, and switches.
Capacity for the G250:
¾
Maximum Extensions
12
¾
Maximum IP Telephones 10
¾
Maximum Local Trunks
4
3.2.7 (3.5.7) DS1/T-1 Carrier Interface Trunk
Vendor Response Requirement:
Provide brief description how DS1-based T-1 carrier trunk circuits are logically and
physically supported by the common control call telephony server, identifying all
intermediary hardware elements necessary for control signaling transmission. Specify
the number of circuit terminations per circuit board/module/media gateway.
Avaya Response:
Avaya G650 Media Gateways
G650 Media Gateways use the TN464GP DS1 Interface, which supports T1 (24
channels) or E1 (32 channels). The circuit pack provides:
¾
Board-level, administrable A- or µ-Law companding
¾
CRC-4 generation and checking (E1 only)
¾
Stratum-3 clock capability
¾
ISDN-PRI T1 or E1 connectivity
¾
Line-out (LO) and line-in (LI) signal leads (unpolarized, balanced pairs)
¾
Support for CO, TIE, DID, and off-premises station (OPS) port types that use
robbed-bit signaling protocol, proprietary bit-oriented signaling (BOS) 24thchannel signaling protocol, or DMI-BOS 24th-channel signaling protocol
¾
Support for Russian incoming ANI
¾
Support for universal, digital, signal level-1equipment in wideband ISDN-PRI
applications
¾
Test-jack access to the DS1 or E1 line and support of the 120A integrated
channel-service unit (ICSU) module
¾
Support for the enhanced maintenance capabilities of the ICSU. These circuit
packs can communicate with Avaya CONVERSANT® or Avaya Interactive
Response.
¾
Downloadable firmware
¾
Support for echo cancellation
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Request for Proposal for an IP Telephony System
Part 1 – Section 3 – Port Interface & Handling Requirements
The echo cancellation capability of the TN464GP is selectable on a per-channel
basis. The TN464GP DS1 interface automatically turns off echo cancellation when it
detects a 2100-Hz phase-reversed tone generated by high-speed modems (56kbps), but not when it detects a 2100-Hz straight tone generated by low-speed
modems (9.6-kbps). Echo cancellation improves a low-speed data call.
The gateway can also use a TN2313AP DS1 port board to interface a DS1 trunk to
the switch backplane via port slots. The TN2313AP supports a variety of
applications, including networking of Avaya switches, international trunk types,
video teleconferencing, and wideband data transmission.
The TN2313AP DS1 interface can be configured as 24 channels at 1.544 Mbps. The
TN2313 can supply two 8-kHz reference signals to the switch backplane for optional
use by the tone clock board in synchronizing the system clock to the received line
clock. The TN2313AP is firmware downloadable.
Avaya G700 Media Gateway
The Avaya MM710 T1/E1 Media Module terminates a T1/E1 connection. The MM710
has a built-in Channel Service Unit (CSU) so that an external CSU is not necessary.
See the following figure for an example of the MM710.
Avaya MM710 T1/E1 Media Module
Highlights of the MM710:
¾
Software selectable T1 or E1 operation
¾
An integrated CSU
¾
Both A-law (E1) and µ-law (T1) gain control and echo cancellation ability
¾
D4, ESF, or CEPT framing
¾
ISDN PRI capability (23B + D or 30B + D)
¾
AMI, ZCS, B8ZS (T1) or HDB3 (E1) line coding
¾
Trunk signaling to support US and international CO or tie trunks
¾
Echo cancellation in either direction
¾
Fractional T1 support
¾
An OIC DB 25-pin interface
¾
A Bantam loopback jack that is used for testing of T1 or E1 circuits.
The MM710 supports the universal DS1 that conforms to the ANSI T1.403 1.544
Mbps T1 standard and to the ITU-T G.703 2.048 Mbps E1 standard.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 3 – Port Interface & Handling Requirements
3.2.8 (3.5.8) 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.
Avaya Response:
All of the media Gateways support trunks described above via included hardware.
The only optional circuit packs that are available work in the G650 Media Gateway
in any available universal slot are:
TN760E tie trunk (4-wire, 4 ports)
The TN760 tie trunk circuit pack has four ports. These ports are used for Type 1 or
Type 5 4-wire E & M lead signaling tie trunks. Trunk types include automatic,
immediate-start, wink-start, and delay-dial. Each port on a TN760 circuit pack has
the following signaling leads:
¾
T
¾
R
¾
T1
¾
R1
¾
E
¾
M.
The TN760 circuit pack provides release link trunks that are required for the
Centralized Attendant Service (CAS) feature and has administrable A- and Mu-Law
companding. The TN760 circuit pack supports outgoing, Multilevel Precedence and
Preemption (MLPP).
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 3 – Port Interface & Handling Requirements
TN763D auxiliary trunk (4 ports)
The TN763 auxiliary trunk has four ports. Each port has the following signal leads:
¾
T
¾
R
¾
SZ
¾
SZ1
¾
S
¾
S
The TN763D circuit pack is used to access on-premises applications such as music
on hold, loudspeaker paging, code calling, and recorded telephone dictation. The
TN763 circuit pack supports external recorded announcement equipment, and is
administrable to select A- or μ-Law companding.
Note: MOH for VoiceCon is provided through analog ports at each location in
this design and no TN763D is required.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 4 - Voice Terminal Instruments
4.0.0. Voice Terminal Instruments
The proposed communications system must be able to support a mix of analog and IP
communications devices. VoiceCon will provide its own analog telephone instruments,
fax terminals, and modems.
Avaya Response:
Read and understood.
4.1
Regulation Requirements
All single- and multi-line IP phones will be manufactured in accordance with Federal
Communication Commission hearing aid compatibility technical standards contained in
Section 68.316. and the Telecommunication Act of 1996.
Vendor Response Requirement:
Confirm the proposed telephone equipment satisfies these requirements
Avaya Response:
Comply. As a responsible telecommunications vendor, Avaya provides equal access
to information because it is the right thing to do. Laws such as the American with
Disabilities Act of 1990, Sections 251 and 255 of the Telecommunications Act of
1996, and Section 508 of the Workforce Investment Act of 1998 guarantee the
right to people with disabilities.
It is very important to recognize that the transmission of TTY tones via VoIP
systems is inherently unreliable. This is true of all current-generation VoIP systems,
regardless of vendor. Although Quality of Service parameters may be adjusted on
VoIP systems, so as to improve the reliability of TTY calls (assuming you’ve been
successful in identifying the TTY calls, which is an entirely different challenge), this
approach does not guarantee reliability. The problem is especially severe when the
Local Area Network (LAN) is congested, or when the TTY calls go out over a Wide
Area Network (WAN). The problem is caused by packet loss.
Specifically, VoIP systems transmit digital audio streams, such as voice and TTY
tones, by breaking the streams into individual packets. Each of these packets is
assigned header information, such as the digital audio encoding scheme that was
used, a sequence number, and a destination. It is important to note that the route
to the destination is not part of the header information. The ability for each packet
to take what is, at that instance, the “best” route to the destination is where VoIP
derives much of its economic advantage. It’s also the reason why TTY-on-VoIP is
unreliable: because packets are free to take different pathways, they cannot be
relied upon to arrive at the receiving device before it’s their “turn” to be played.
Although these packets often arrive eventually, they are regarded as lost because
they did not arrive in time, and must therefore be discarded. Under most
circumstances, the loss of occasional packets is not detectable in voice
communication. The reason is that VoIP telephones employ packet loss
concealment algorithms that trick the human ear by mimicking the acoustic
properties of adjacent packets. Although these techniques work well with voice,
they do not work with TTY tones.
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Part 1 – Section 4 - Voice Terminal Instruments
If a packet containing a TTY tone is lost, the current generation of VoIP packet loss
concealment techniques is unable to recover it or rebuild it.
As an indication of the severity of this problem, with VoIP systems that employ a
packet size of 20 milliseconds (a typical value), the TTY character error rate will
exceed the FCC’s recommended limit of one percent when the packet loss rate
exceeds 0.12% -- a rate generally regarded as excellent for voice communication.
TTY-on-IP - Packet loss in an IP network of greater than 0.12% renders a TTY
device unusable when the voice channel is used for transmitting the TTY's audio
tones. In addition, audio compression schemes such as G.729 do not always have
the ability to capture the tones' frequency and/or duration with adequate precision.
One workaround for audio compression deficiencies is to use G.711; however, this
requires 8 times as much bandwidth, which can seriously impact the WAN costs.
Avaya has made tremendous strides in solving these technical issues with the
introduction of TTY-on-IP in Avaya Communication Manager. The solution transmits
a verbal description of the tone (rather than the tone itself) via a data channel that
incorporates the ability to retransmit missing packets. This approach can tolerate
considerably higher packet loss than typical solutions and works with all codec
compression schemes. In fact our TTY-on-IP solution permits voice and TTY to be
mixed on the same call, something that is not possible with solutions offering
"instant messaging" as a substitute for TTY.
This feature enables Avaya to comply with Titles II, III, and IV of the Americans
with Disabilities Act of 1990, Sections 251 and 255 of the Telecommunications Act
of 1996, and Section 508 of the Workforce Investment Act of 1998 for a pure IP
solution.
Avaya can also address the issue of TTY compatibility in VoIP systems by
supporting a hybrid architecture. Specifically, the Avaya solution is able to provide
error-free transport of TTY tones because, in addition to VoIP, Communication
Manager easily supports TTY-compatible analog and TDM digital ports. Because of
this hybrid architecture, the solution can ensure that all TTY traffic is transmitted
via TTY-compliant non-VoIP pathways, while still permitting voice calls to receive
the full benefits of VoIP technology.
With regard to Section 255, it is very important to note that the Federal
Communications Commission does not assert jurisdiction over Voice over Internet
Protocol systems. (Because these systems rely on Internet mechanisms for voice
transport, the FCC has classified them as information systems, rather than
communication systems.) For this reason, under the law, VoIP systems are exempt
from the requirements of Section 255. This means that a VoIP manufacturer could
claim legitimately that they are not out of compliance with Section 255, even if
their system failed to satisfy even the most basic needs of disabled people. We do
not claim this exemption for our VoIP systems.
December 1, 2006
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Part 1 – Section 4 - Voice Terminal Instruments
With regard to Section 508, an exemption is permitted for devices that are located
in areas frequented only by service personnel for the occasional monitoring or
maintenance of equipment. Although the intent of the regulation was to relieve
organizations from having to make their equipment closets wheelchair-accessible,
some vendors have stated that this exemption means that their switches and
routers do not need to comply with Section 508’s telecommunications
requirements. We do not claim this exemption for our switches and routers.
Unified Messaging for TTY users
Avaya Modular Messaging also allows callers to select whether they wish to be
prompted by voice or in TTY format. In addition to leaving messages, TTY users can
take advantage of an Avaya messaging system interface to log in and manage their
mailboxes via TTY.
Access for Visually Impaired Users
Avaya has developed a PC-based application called Universal Access Phone Status
that helps visually impaired users manage voice communications in their
Communication Manager environments. Avaya telephones have over 200 different
functions that may be displayed visually. Functions such as caller ID, message
waiting lamp, which lines are on hold, what lines are available and if "Send all calls"
is activated present challenges for the visually impaired until now. This software
package also has great applicability in a call center environment where a visually
impaired agent can function as a productive member of the split.
The Universal Access Phone Status application (referred to hereafter as APS)
provides information about a user’s telephone display and lamp status through the
PC’s speakers or monitor. The APS application allows the user to choose between
audio and visual notification for specific telephone events such as an incoming call.
The capability provided is determined by the type of notification you set up when
you configure the application. For users who select the Visual notification mode, the
primary function of the application is to provide a visual alert on the PC monitor
when an incoming call occurs. When you configure a user for the Audio notification
mode, buttons on that user’s telephone that are unassigned on the telephone
switch can be configured in APS to perform four functions:
¾
Play Contents of Phone Display
¾
Play Phone Lamp Status
¾
Repeat Last Playback
¾
Stop Playback Immediately
Universal Access Phone Status is available at no extra cost to customers using
Avaya Communication Manager, Release 1.3 and later.
December 1, 2006
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Part 1 – Section 4 - Voice Terminal Instruments
4.2
Desktop IP Telephone Instruments
VoiceCon has a requirement for several types of desktop IP telephone instruments:
•
Economy
•
Administrative
•
Professional
•
Executive
Avaya Response:
Comply. The proposed design includes native IP telephones for all instruments
specified in this requirement, except the analog phones. We have included Avaya
Softconsole application licenses for the required attendant positions. The IP models
included are:
Economy
Avaya 4601 IP Telephones
Administrative Avaya 9650 IP Telephones
Professional
Avaya 9630 IP Telephones
Executive
Avaya 9640 IP Telephones
All of the proposed IP telephones support IEEE 802.3af standard compliant Power
over Ethernet (PoE).
4.2.1 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 illustration/ photograph (PPT format,
only) of the instrument. Indicate in your response any and all requirements not satisfied.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 4 - Voice Terminal Instruments
Avaya Response:
Comply. The proposed solution includes Avaya 4601 IP Telephones. The Avaya
4601 is a cost effective, entry-level IP telephone with 2 call appearances. Please
note that since this telephone does not have a display, display based capabilities
such as the push feature described elsewhere in this proposal do not apply. A
picture of the Avaya 4601 IP Telephone is included in Appendix 3, PowerPoint
Illustrations.
The following are characteristics of an Avaya 4601 IP telephone:
¾
2 call appearances with LEDs
¾
Fixed button with LED for voice mail retrieval
¾
Five fixed feature buttons including:
o
Hold
o
Transfer
o
Conference
o
Drop
o
Redial
¾
Supports 802.3af standard power over Ethernet
¾
Supports Quality-of-Service features including RTCP and RSVP
¾
Wall or desk mount
¾
10/100Base-T Ethernet network connection with RJ-45 interface
¾
IP address assignment using DHCP
¾
Downloadable firmware for future upgrades
¾
12-button touch-tone dial pad with raised bar on the button labeled five for the
visually impaired
¾
Message waiting light (LED)
¾
Hearing aid compatible
¾
Adjustable volume control
¾
Available in dark gray
¾
Available world wide
¾
Supports G.711, G.729A, and G.729B audio voice CODECs
¾
Supports H.323 V2
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 4 - Voice Terminal Instruments
4.2.2 Administrative Desktop IP Telephone Instrument
The Administrative model will be used by station users who have executive management
group call answering and coverage responsibilities. It should have, at minimum the
following design attributes and features/functions:
•
12 key dial pad
•
Sixteen (16) programmable line/feature keys with soft label/status indicators
•
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 greyscale pixel-based, graphical display
screen with four (4) associated context sensitive soft keys
•
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 if 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.
Avaya Response:
Comply. The Avaya 9650 IP Telephone exceeds the requirements listed. Please see
Avaya Appendix-3-PPT-Slides.ppt for photos.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 4 - Voice Terminal Instruments
The 9650 is designed to meet the needs of an Administrative Assistant or
Receptionist for improved call handling and call management; it supports 24 call
appearance/CM features and can be used with the optional SBM24. The Avaya 8650
IP telephone pprovides 11 one touch buttons for quick access to Communication
Manager Features or Call Appearances
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 greyscale pixel-based, graphical display
screen with four (4) associated context sensitive soft keys
•
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
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 4 - Voice Terminal Instruments
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.
Avaya Response:
Comply. The Avaya 9630 IP Telephone exceeds the requirements listed by adding
an optional Avaya ABT35 Bluetooth Headset
Please see Avaya Appendix-3-PPT-Slides.ppt for photos.
A member of the Avaya one-X Deskphone Edition family, the 9630 IP Telephone is
specifically designed for the essential telephone user—those for whom the
telephone is essential in order for them to complete their job. Sales people,
relationship managers, and attorneys are typical examples of the essential users’
profile. The 9630 provides superior, high fidelity audio, built in "one touch" access
to key Avaya Communication Manager mobility features, and a stylish and
professional design.
Product Details
The 9630 IP Telephone features a 3.8 inch (9.65 cm) diagonal monochrome backlit
display - which has been enhanced with higher resolution (1/4 VGA) versus other
available monochrome telephones from Avaya. The 9630 supports up to 24 call
appearances / administered feature keys - with six concurrent line appearances
visible at any time.
The 9630 has several LED buttons throughout the front of the phone. Six LED line
appearance buttons on the side of the display provide explicit status of different line
appearances and administered features, while LEDs built into several buttons on the
phone such as Mute, Message, and Headset provide an intuitive and simple
experience for the everyday end user.
The user interface on the 9630 is helpful and intuitive. So completing call transfers
and setting up ad hoc conference calls is simple and can be executed with
confidence. The 9630 has a dedicated Call Forward / Mobility button - which
provides direct access to Communication Manager Mobility features such as
Extension to Cellular and Extend Current Call - features critical to the essential
user.
The 9630 supports an optional 24 button expansion module. This provides the
essential user with additional call appearances, bridged appearances and
administered feature keys including speed dials.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 4 - Voice Terminal Instruments
The 9630 requires electricity from either an 802.af Power over Ethernet switch or a
local Avaya power supply. The 9630 is a Class 2 PoE device. Additionally, when
configured with an SBM24 button expansion module, the 9630 supplies power to
the SBM24 - via the module interface on the phone. No separate power supply is
required for the SBM24. Even in this configuration, the 9630 plus SBM24 are still
PoE class 2.
ABT-35s Bluetooth Headset
The optional ABT-35Ss is a comprehensive wireless headset system designed to
seamlessly support the communications needs of today’s business traveler. Based
on Bluetooth wireless technology, the ABT-35s headset and base unit supports the
Avaya desktop phone as well as any Bluetooth compatible cell phone or handheld
device. With this single, lightweight headset, users will experience the freedom,
convenience and comfort of hands free voice communications whether in the office
or on the go.
In the office, the ABT-35S base unit attaches to the headset / handset jack of the
Avaya desktop telephone (both Avaya IP and Digital phones are supported). The
headset allows for wireless freedom with clear voice quality up to 33 feet (10
meters) from the base. In addition, the headset itself supports alerting and switch
hook control of the Avaya desk set – allowing users to answer incoming calls and
carry out conversations – even when away from their desks.
FEATURES AND BENEFITS
Productivity of Users - Through prompting for common telephony tasks, one touch
access to key features, and superior high fidelity audio - the productivity of end
users is greatly enhanced.
Richer Communication - The superior audio capabilities make conference calls and
meetings more effective by requiring less reiteration. This has been found to reduce
employee stress and fatigue.
Investment Protection - Built on open standards with a modular platform that
supports a wide range of modules and adapters to further enhance user
productivity, provides investment protection.
December 1, 2006
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Request for Proposal for an IP Telephony System
Part 1 – Section 4 - Voice Terminal Instruments
Key Features
Benefits
Intuitive User Interface
The intuitive, context-sensitive
interface on the 9630 is designed to
facilitate confident usage by featuring
context-driven menus with on-screen
prompts, enabling straightforward
access to contact directory and call log.
Essential users have full control of
conference calls, including selective
drop and mute. In addition, the 9630
provides a dedicated Call Forward /
Mobility button - for "one touch"
access to mobility features within
Avaya Communication Manager. These
include Extension to Cellular as well as
Extend Current Call to cell phone.
Superior Audio Quality
The unique high-fidelity acoustics of
the Avaya 9630, including wideband
audio support in the speaker, handset,
and headset deliver industry-leading
audio that minimizes ambient noise.
With the enhanced audio across high
and low frequencies, it is easier for
users to better understand others with
different speech nuances or accents.
New design and display
The 9630 features an improved, higher
resolution display - supporting 1/4 VGA
gray scale with backlighting. A four
way navigation button cluster is
another new addition to the 9630 providing a familiar, cell phone-like
interface for navigation and feature
selections for the everyday user.
December 1, 2006
Easy access to common features such
as Conference, Transfer, Hold are now
greatly enhanced with the helpful
prompts and tight integration with
phone numbers in the contact list and
call logs. The user interface on the 9630
provides easy access to critical
Communication Manager mobility
features, allowing them to be reached
transparently whether at their desk or
on the go.
For the essential end user, the person
constantly on the telephone the
enhanced audio allows calls to be more
productive with team members better
able to collaborate. On conference calls,
users find it easier to distinguish and
understand multiple speakers, aiding in
collaboration and communications.
Overall, communications are richer.
The new design facilitates better usage
of the display and the built-in browser
to improve access to information and
use of telephone features.
©2006 Avaya Inc.
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Request for Proposal for an IP Telephony System
Part 1 – Section 4 - Voice Terminal Instruments
Key Features
Benefits
Investment Protection - Modules
and Adapters
Based on open standards with a
modular platform, the Avaya 9630
allows enterprises to add a wide range
of modules and adapters to further
enhance employee productivity. This
facilitates individual choice between
Bluetooth, monaural - wideband or
binaural - wideband headsets. A
standard USB interface accommodates
a range of USB devices. In the future,
additional adapters and modules
including those for Gigabit Ethernet
and Bluetooth will be available.
9630 Designed just for the
essential user
The 9630 is designed specifically for
the essential user - for whom the
telephone is critical for them to
perform their jobs. The 9630 features
a larger display, with additional LED
buttons and wideband audio support
throughout. The 9630 supports a 24
button expansion module - allowing
the essential user to access additional
call appearances and feature keys.
Finally, with one touch access to
mobility features, the 9630 user is
reachable seamlessly.
Investment protection and enhanced
total cost of ownership are the key
benefits. Traditionally, the only way to
receive new telephone capabilities - was
to purchase a completely new phone.
Designed with the future in mind, the
Avaya 9630 provides a flexible approach
for adding future functionality to current
telephones.
Having a match between the
communication needs of the essential
end user and the phone on their desk
provides a business with great
advantages -- especially given the
importance of the roles held by essential
users (sales, customer facing
associates). Allowing the 9630 user to
make effective use of the right set of
telephone features confidently, enables
them to perform their job more
proficiently, and ultimately helps to
satisfy customers and provide
competitive advantages.
SECURITY
Security Features
Denial of Service Protection. The 9600 Series IP Telephones are designed to
continue to function, even during active calls in the event of a denial of service
attack.
Authenticate downloaded software. The 9600 Series telephones authenticate all
downloaded software via a digital signature to ensure that only authentic Avaya
software is ever installed in the telephone.
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802.1X support - forwarding and supplicant. The 9600 Series telephones support
several modes of 802.1X operation that include supplicant operation for true
authentication of the telephone, pass-through of 802.1X messages for
authentication of an attached PC, and a multi-supplicant mode in which both the
telephone and the PC can be authenticated.
Proxy-logoff. Proxy-logoff is also supported to alert the network if an authenticated
PC is disconnected from the telephone.
VLAN separation. The 9600 Series telephones support VLAN separation to ensure
that traffic from an attached PC cannot get onto the voice VLAN, and to ensure that
any traffic on the data VLAN does not affect telephone operation or voice quality.
The VLAN ID and QoS priority of frames transmitted by an attached PC can be "remarked" with specific administered values.
FUNCTIONAL ATTRIBUTES
High Fidelity Audio
The audio quality on the Avaya 9630 Telephone sets a new standard in phonebased communications—you’ll have to hear it to believe it. Based on the latest VoIP
chip technology, Avaya has changed the acoustic design of the phone body,
speaker and handset to reduce background noise and improve audio quality. As a
result, participants on calls can better hear each other – eliminating repetition,
reducing fatigue and improving overall communication.
With support, for the open standards based G.722 wideband audio codec, along
with Avaya Communication Manager integration, you are now able to hear a
broader range of audio frequencies you might not have known you were missing—
until now.
User interface simplified and enhanced
The 9630 is designed to make critical telephone functions easier and faster to
accomplish. Take the case of transferring a call: How often do you need to “remind”
yourself exactly how to do this routine activity? How often do you also provide the
extension to your connected party, “in case you lose them.” The 9630 IP telephone
continues to make common functions like Hold, Conference and Transfer accessible
via softkeys on the display itself only now with enhanced functionality.
Timely screen prompts and scroll down menus now guide you step by step through
the process. Making key features easy to access helps everyone get things done
more quickly and confidently. Display options for conferencing, mobility, directory,
and access to third party applications present a whole new realm of
communications. For example, third party appli-cations deliver Microsoft Outlook
capabilities right from the display of the phone—reminding you of an upcoming
conference call and providing the conference bridge number.
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Cutting edge and sleek architectural design
Phone features have evolved over the years, but a com-plete overhaul of phone
design and user interface creates a new opportunity for enhanced end user
experience. The Avaya 9630 IP Telephone has been completely re-engineered
specifically to address the needs of the essential user. The result is a professional
and timeless design that delivers a new level of functionality and efficiency within
while providing visual appeal throughout.
Investment Protection and flexibility for the future
The 9630 is designed for future growth and enhancement. The Adapter interface on
the backs of the phone provides support for additional wired and wireless network
environments. Beyond Gigabit Ethernet, adapters for 802.11 WiFi and Bluetooth
wireless networks will be available in the future. The Module interface on the 9630
will support a growing list of devices integrated at each desktop. The Modular
design of all the 9600 Series telephones including the 9630, allows customers to
leverage initial phone investments with enhanced capabilities as needed in the
future. This flexibility provides enhanced total cost of ownership.
The Avaya 9620, 9630, 9640, and 9650 IP Telephones, support local call log, local
speed dial and web browser applications, which provides access to web-based
applications and information as well as the added efficiencies of speed dials and call
logs stored in the telephone.
Push of Content
The 96xx Series IP Telephone firmware (which applies to the 9620, 9630, 9640,
and 9650 IP telephones), provides support for a feature called "push". "Push" is the
capability of an application using the WML protocol to send content to the telephone
without the user requesting it, and (potentially) overriding what the user is
otherwise experiencing. This capability is available to 3rd-party developers through
Software Developer Kits (SDKs). Content can be pushed to the phones via a text
message, WML web page, or an audio stream through the handset or speaker.
IP telephones become applications-enabled appliances instead of just phones. The
addition of push enables phone displays and/or audio to support a variety of
applications (web browsing, time reporting, emergency alerts, travel reservations,
account code entry, announcements, branding via screensaver, inventory lookups,
scheduling, etc).
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Avaya Phone Application Suite (Optional)
Supported on the 9620, 9630, 9640, and 9650 IP telephones, the Avaya Phone
Application Suite consists of a pre-packaged set of productivity enhancing phone
applications. These applications are easy to deploy and leverage the large, bright
displays and embedded micro browsers on the Avaya IP telephones.
The Avaya Phone Productivity Pack includes the following three applications:
¾
Broadcast Server: allows for text, graphics, and audio message broadcast to
the displays and speakers of IP phones within a location. Great for simple
reminders or alert messages such as an email server being taken temporarily
off line
¾
Text Messaging: provides simple SMS text messages between IP phone users
¾
Backlit and color display options
¾
Depending on the model, the 46xx series of IP phones provides a number of
different backlit and color display options.
The 9630, 9650 include a user adjustable backlit display. The 9640 supports highdensity color displays.
With large, bright and colorful displays, the 96xx series of IP phones are designed
and optimized for the delivery of content and applications directly to the phone.
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 keys
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•
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 Ethernet connection
•
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.
Avaya Response:
Comply, the Avaya 9640 IP terminal meets or exceeds the requirements and same
capabilities as the 9630 listed above, The 9640 does have a High resolution, backlit,
color pixel-based, graphical display screen as requested.
Please see Avaya Appendix-3-PPT-Slides.ppt for photos.
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 IP desktop telephone instrument model. 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).
Avaya Response:
Comply. The web browser application provided with the presented Avaya 96xx
Series IP Telephones is intended to be a relatively simple interface that can be used
to access Wireless Markup Language (WML) based web information. VoiceCon may
use off-the-shelf WML web authoring tools for intranet web sites. If administered, a
URL will be downloaded to the telephone that is used as the “home” page for this
application. Navigation is via activation of links on the home page (and on
subsequent pages), and via Back, Forward, Home, Refresh and scrolling controls.
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The line-based interface and 4-grayscale display provide an interface similar to that
of a Wireless Application Protocol (WAP) telephone or a Personal Data Assistant
(PDA) such as an early Palm Pilot. For that reason, the Avaya IP Telephones use
Wireless Markup Language (WML) to interact with an external web server. The IP
telephones convert the WML sent to it by this external server, and pass softkey and
feature button presses back to the server as appropriate.
The initial version of the IP telephone browsers supports WML 1.3. The browser
supports all required tags, but is not able to display a table or render any images.
Proprietary tags are not supported at this time. The browser makes use of the
Feature Buttons on the phone and softkeys for navigation on the display as well as
to other connected pages.
The IP telephones do not have a mouse to navigate around the screen, so the
Feature Buttons that are located on the left side of the screen are used to select a
particular line on the display, or "to bring that line into focus". Focusing on a line is
used to select a line for text entry or to select a line that contains a link to another
URL.
The Avaya 96xx product family also supports the ability to push textual content to
the displays and audio content to the speakers, of the 9600 Series IP telephones.
"Push" is the capability of an application using the WML protocol to send content to
the telephone without the user requesting it, and (potentially) overriding what the
user is otherwise experiencing. This optional capability has been made available to
third-party developers through a Software Developer Kit (SDK), known as the SDK
for IP Telephones 2.1, posted to support.avaya.com . Content can be pushed to the
phones via a text message, WML web page, or an audio stream through the
handset or speaker.
As our IP telephones evolve into applications-enabled appliances instead of just
phones. The addition of “push” enables phone displays and/or audio to support a
variety of applications (emergency alerts, visual voice mail, announcements,
branding via screensavers, zone paging, inventory lookups, scheduling, etc) and
enhances existing applications that are accessible via the phone display.
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Attributes Beyond the Scope of the Requirements
In addition, several Avaya IP Telephones provide four interfaces to communicate
with the telephone.
Browser Interface
TAPI: This is a Telephony Application Programmer's Interface. TAPI is primarily
used for call control applications such as NetMeeting or IP Softphone.
Web: This is a Web Browser Interface. Users can navigate web applications and
retrieve information about the company, news, interactive applications such
as the conference room scheduler, Company Directory lookup, etc.
Push: This is an optional interface available with Release 2.1 software of the Avaya
IP Telephones. Until now, external applications used to interface with the IP
telephone using the Web Browser. The Web Browser Interface requires users
to "Pull" the data from an application, i.e., a user has to click on a particular
link or page to access the information. With the Push Interface, the
application can itself spontaneously "Push" the information to the telephone
without the user having to click on a link.
Some of the examples of an application for the “Push” capability are:
¾
Broadcasting company news
¾
Sending meeting reminders with conference bridge numbers so that users
don't have to search for the conference number
¾
Streaming music such as wake-up alarms in hotel rooms
¾
Streaming announcements for academic environments such as college
campuses
¾
Sending critical stock news information to the stock broker's telephone
¾
Broadcasting critical weather alerts to employees
¾
Building intelligent databases to target information to an individual or groups
of phones.
Please note – the “Push” application is an optional feature
4.2.5 Desktop Instrument Options and Add-on Modules
Vendor Response Requirement:
Provide a brief description of available hardware/software options and/or add-on
modules available with the proposed Economy, Administrative, Professional, and
Executive models. Options/modules should include key modules, display modules,
Gigabit Ethernet connectors, and et al. necessary to satisfy the above telephone model
requirements. Indicate the specific models that support the listed option/module.
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Avaya Response:
Avaya Gigabit Adapter for 9600 Series (Optional)
¾ PoE and local power, Class 3
¾ Adapter provides investment protection, migrate at your own speed
¾ Integrated Gigabit ports support green fields and infrastructure-in-place
customers
¾ Compatible with all 9600 Series IP Phones that have an adapter interface
¾ Requires the use of the Wedge Stand (sold separately)
Integrated Gigabit Adapter for 9600 Series Phones (not actual size)
Integrated Gigabit Adapter for 9600 Series Phones installed in a 9630 shown with
Wedge Stand (bottom view)
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Avaya Phone Application Suite (Optional)
Optionally, Avaya has adjuncts and third party applications that provide more
robust functionality, such as access to the end users outlook calendar, video,
broadcasting of a variety of message and screens, as well as other industry specific
applications.
The Avaya Phone Application Suite creates a new communication and application
paradigm for Avaya IP telephone users. Beyond the hard dollar cost savings
associated with IP telephony, with the Avaya Phone Application Suite, the 4600
series IP telephones become multi-function, intelligent endpoints on the IP network
- capable of running productivity enhancing applications outside the context of a
telephone call.
The initial release of the Avaya Phone Application Suite provides the AG250 Phone
Application Gateway server along with Broadcast Server and Design Studio
applications. The Broadcast application is supported on the Avaya 9630 9640 and
9650 IP Telephones.
Components
The Avaya Phone Application Suite consists of several components. Broadcast
Server is a powerful application allowing for text messages, graphics and audio to
be pushed out to Avaya IP phones.
Design Studio is a software tool that allows existing web based applications to be
transformed and made accessible right from 9600 series IP phones. The AG250
Phone Application Gateway is a rack mountable server, which integrates with Avaya
Communication Manager and allows for applications to be accessed from Avaya IP
telephones. With implementation and maintenance support from the qualified
engineers within Avaya Global Services, the Phone Application Suite is a complete
solution for desktop productivity.
Avaya AG250 Phone Application Gateway
Broadcast Server
The Broadcast Server application leverages a customers IP infrastructure and 9600
Series IP telephones by allowing an administrator to broadcast audio, text, and
images to the displays and speakers of the phones. This allows for simple
communications to reach a large number of workers instantly. Say for example, the
IT department needs to take a server off line momentarily, an administrator could
send a broadcast which plays a horn sound through the phone speaker, then
displays the notification about the server going off line. Unlike email, the Broadcast
Server application is ideal for cutting through all the clutter and gets information
communicated quickly and efficiently. Broadcast messages can be composed and
delivered on the fly in an ad hoc manner, or can be easily created and scheduled for
future delivery. This is useful in scenarios where simple reminder messages are
required or helpful.
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Text Messaging
The Text messaging application allows workers to both send and receive simple text
messages utilizing their IP telephones. Users can pick from pre-created, simple
messages such as "running late" or "I'm leaving now", or messages can be
composed using the dial pad of the phone and then sent to one or multiple
recipients. Messages can only be sent to other IP phones. This application is
designed for use between devices behind VoiceCon’s firewall.
Broadcast Server: distribution lists
The Broadcast Server application supports a browser based interface for the
administrator to compose and broadcast messages out to Avaya IP telephones. The
Broadcast Server supports the use of distribution lists for the sending of messages.
Like with email or voicemail, distribution lists make broadcasting of information
easier and more efficient.
Text messaging: support for multiple recipients
The Text Messaging application supports the sending of text messages to multiple
recipients simultaneously. So for example, in the case of a person running late to
join a meeting, they could quickly compose and send a message for the meeting
organizer - going instantly to the organizers desk phone and conference room
phone. This enhances the ability for workers to collaborate and keep in touch with
each other real time.
Transformation of web applications for use on IP Phones
The Design Studio software tool allows for existing web based (HTML, XML)
applications to be transformed for use on Avaya IP telephones. Existing web based
applications can be easily leveraged and extended to IP phones for access. Avaya
Global Services has available a per day Professional Services offer for custom
deployments based on Design Studio.
Enhanced Worker Productivity
The Avaya Phone Application Suite allows workers to enhance their productivity by
running value added applications right from the displays of their Avaya 9600 Series
IP telephones. Worker productivity increases, the ability to communicate and
collaborate with others increases as well.
Total Solution from Avaya
Avaya Global Services is available for up front planning, consultation,
implementation and maintenance support. VoiceCon benefits from looking to Avaya
- a single, accountable solution provider for the Phone Application Suite, for all
support needs.
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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, 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 any
proposed models that cannot currently be programmed for SIP support based on current
commercial availability standards.
Avaya Response:
The 9600 Series telephones can be configured to support either H.323 protocol, for
integration with traditional Avaya IP Telephony Servers and Gateways, or SIP
protocol, for support of SIP Communications Servers such as the Avaya SIP
Enablement Services (SES) Release 3.0 Solution. The signaling protocol can be
toggled from the dial pad via a local administrative code (after which the
instrument will acquire the alternative firmware from the local HTTPS/HTTP server).
Avaya SIP Enablement Services (SES) creates a communication services layer
within the Avaya Communications Architecture that mediates between Avaya
MultiVantage applications and a wide range of standards-based user agents, webbased applications, and communication devices. These services combine the
standard functions of a SIP proxy/registrar server with SIP Trunking support and
duplicated server features to create a highly scalable, highly reliable SIP
communications network. This resulting network supports telephony, instant
messaging, conferencing, and collaboration solutions.
Avaya SIP Enablement Services provide a compelling value proposition that allows
enterprises to maximize economic and productivity gains through the
implementation of SIP interoperability and capabilities, with an evolutionary path to
migration that fully protects existing investments and minimizes service and
business interruption.
SIP Trunking
Enterprises can start by taking advantage of new Service Provider IP SIP Trunking
service. This is as simple as adding SIP Enablement Services to their existing
Communication Manager (3.0 or above) network. Trunk support conforms to
applicable SIP standards and the SIPConnect specification, an industry initiative to
define a standards-based approach to direct IP peering between SIP-enabled IP
PBXs and VoIP service provider networks. This allows the enterprise to take
advantage of SIP-based PSTN origination/termination services offered at very
competitive prices, without making any other changes to their existing system.
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Instant Messaging (IM)/Presence
Once SIP Enablement Services are in place, a gradual step-by-step migration path
allows the enterprise to maximize the benefits of SIP while fully preserving
compatibility and investment protection with existing H.323, digital and analog
endpoints and infrastructure. For example, SIP Enablement Services can be
leveraged to introduce secure Enterprise Instant Messaging (IM) and user presence,
integrated with telephony, through the Avaya IP Softphone and IP Agent client
applications. IP Softphone and IP Agent combine H.323 telephony support with a
SIP-based IM client and a presence-enabled contact directory that supports both
voice and IM communications. This allows enterprises to extend the benefits of user
presence and IM to all users without the need to make extensive changes to their
existing voice infrastructure.
SIP Telephony
SIP supports both numerical (telephony) and alphanumeric addressing, providing a
critical bridge for communications between PSTN and Internet networks. This allows
users on either network to reach any other user without giving up existing devices
or the advantages of each. When enterprises are ready to make the transition to
SIP Telephony, Avaya SIP Enablement Services will allow them fully leverage the
benefits of SIP, while supporting full compatibility and use of their existing
endpoints.
For enterprises that have already deployed H.323 IP Telephony, the migration
process can be initiated immediately and transitioned at whatever pace is desired.
Once registered and licensed on SIP Enablement Services, existing Avaya IP phones
can convert their operation from H.323 to SIP through a simple and free firmware
upgrade. Through Communication Manager Extended Access, SIP endpoints have
access to additional telephony features. IP Softphone users have a similar migration
path to SIP telephony through the Avaya SIP Softphone, which supports SIP for
both IM and telephony.
A key feature of SIP is its support of Uniform Resource Indicators (URI) for user
addressing, in the same basic form as e-mail addresses (i.e. [email protected]).
Because this addressing is based on the user, not a device, it can be mapped to
whatever device the user desires. Ultimately, this will allow people to communicate
with each other using a single handle-based address vs. the hard-to-remember
multiple phone numbers of their desk phone, cell phone, pager, etc. Through SIP
Enablement Services, communication over even existing telephony networks
become simpler, more intuitive, and focused on the user vs. a device.
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User Mobility
SIP is well-suited for mobility requirements. When a user logs onto a SIP device, it
registers the user and sends the SIP URI of the device to the registrar service,
which is used to route calls to/from the user. Avaya Personal Profile Manager, a
user mobility application within SIP Enablement Services, leverages this capability
to create a custom work environment that follows the user. Personal Profile
Manager is centralized service that resides in each SIP Home Proxy Server and
communicates with its assigned SIP endpoints to receive, store, and distribute user
profile information.
SIP Applications
For Avaya, SIP is a catalyst for the evolution of enterprise communications to
Converged Communications, as telephony migrates from a standalone
communication solution to a multi-modal communication core service that can be
integrated with other business applications. Avaya is taking the lead in the
modularization of our software and systems into open communication architecture.
As solutions become more modular, their services can be deployed in a greater
number of configurations and more easily integrated within a multi-vendor
environment.
Through SIP Enablement Services, a number of Avaya MultiVantage™ Business
Communication Applications, including Communication Manager Telephony and
Meeting Exchange web/audio conferencing, become available to a wide range of
standards-based user agents, web-based applications, and communication devices
to create a new paradigm of Converged Communications that will lead to increased
flexibility and cost efficiency.
SIP Standards Supported
Avaya’s SIP solutions support the following IETF standards for SIP:
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
RFC 1889 RTP: Real-Time Transport Protocol
RFC 2246 The TLS Protocol
RFC 2327 SDP: Session Description Protocol
RFC 2396 URI generic syntax
RFC 2617 Digest Authentication
RFC 2782 A DNS RR for specifying the location of services (DNS SRV)
RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony
Signals
RFC 3261 SIP: Session Initiation Protocol
RFC 3262 Reliability of Provisional Responses in the Session Initiation
Protocol (SIP)
RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers
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¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
RFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP)
RFC 3265 SIP-Specific Event Notification: Message Summary and Message
Waiting Indication Event Package
RFC 3311 UPDATE method (partial support)
RFC 3325 Private Extensions to the Session Initiation Protocol (SIP) for
Asserted Identity within Trusted Networks
RFC 3420 Internet Media Type message/sipfrag
RFC 3428 Session Initiation Protocol (SIP) Extension for Instant Messaging
RFC 3515 REFER method
RFC 3551 RTP Profile for Audio and Video Conferences with Minimal Control
RFC 3578 Mapping of Integrated Services Digital Network (ISDN) User Part
(ISUP) Overlap Signaling to the Session Initiation Protocol (SIP)
RFC 3840 Indicating User Agent Capabilities in the Session Initiation Protocol
(SIP) (partial support)
RFC 3841 Caller Preferences for the Session Initiation Protocol (SIP) (partial
support)
RFC 3842 A Message Summary and Message Waiting Indication Event
Package for the Session Initiation Protocol (SIP)
RFC 3856 A Presence Event Package for the Session Initiation Protocol (SIP)
RFC 3891 The Session Initiation Protocol (SIP) "Replaces" Header
draft-elwell-sipping-redirection-reason-00
draft-ietf-impp-cpim-03
draft-ietf-impp-cpim-pidf-07
draft-ietf-simple-winfo-format-04
draft-ietf-simple-winfo-package-04
draft-ietf-simple-presencelist-package-00
draft-ietf-sip-history-info-03
draft-ietf-sip-join-03
draft-ietf-sip-session-timer-15
draft-ietf-sipping-cc-conferencing-04
draft-ietf-sipping-cc-transfer-02
draft-ietf-sipping-conference-package-01
draft-ietf-sipping-dialog-package-04
draft-ietf-sipping-realtimefax-01
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4.3
PC Client Softphone
A PC client softphone will be used by station users and attendant operators as their
primary desktop voice terminal. The PC client softphone application should conform to
SIP standards and specifications.
Vendor Response Requirement
Confirm that the proposed softphone solution satisfies the stated SIP requirement.
Avaya Response:
Comply.
Avaya IP Softphone
IP Softphones allow you to access the features of Avaya Communication Manager
without having to be tied to one location. One of the main benefits of IP Softphones
is that you can load them on a laptop PC and connect them to the switch from
almost anywhere through a network connection.
IP Softphone Release 5.1 Features
Enhanced desktop integration via click-to-dial from Internet Explorer (5.0 and
higher) page and name look-ups and dialing from Microsoft Outlook Contact lists
and LDAP directories
Improved user productivity via shared control of DCP telephones (6400 and 2400
series only and requires Avaya Communications Manager 2.0 or higher)
The Avaya IP Softphone allows you to easily make and receive calls by Voice-overIP (VoIP), from any location, using a simple graphical user interface on your PC or
laptop computer screen. You can even select a graphical interface that looks just
like a desktop phone.
An illustration of the Avaya IP Softphone is included in Appendix 3, PowerPoint
Illustrations.
For employees who work out of the main office—on the road or at home—the Avaya
IP Softphone creates one simple, easy-access method for accessing the office
telephone. Once the Softphone software is installed on your PC or laptop, you can
place and receive calls—and access voice mail messages—using high-quality iClarity
IP Audio. You can also access the powerful Avaya Communication Manager station
features such as multiple call appearances, caller ID, conference, as well as speed
dial, send all calls, and message waiting.
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Avaya IP Softphones can have one of three configuration types: Road Warrior,
Telecommuter, or IP Telephone. The Road Warrior configuration is suitable for users
who tend to have only one line for remote access. In Road Warrior, the voice or
audio is run across the IP network for a "pure voice over IP" configuration, and
offers a great amount of flexibility due to the ubiquity of IP networking. The
Telecommuter configuration is ideally suited for users working from a remote office
with two lines for remote access. With this option, feature/access control and
signaling is maintained and delivered across the IP network, but the voice is
delivered across a second line to either a public switched telephone network or
digital line to help ensure toll-quality voice. This capability can be extended to a
cellular, PCS, or GSM phone. In the Telecommuter configuration, the Avaya
communications server "binds" the two connections as a single transaction or
session.
Avaya IP Telephone configuration enables users to log into and control their Avaya
IP Telephone from the Avaya IP Softphone. Users can speak and listen through
their telephone, but unlike the Telecommuter configuration, users can make and
handle calls from both the Avaya IP Softphone interface and the IP Telephone. This
feature can help improve productivity by integrating the IP Softphone capabilities
and Personal Information Managers, such as Microsoft Outlook, with the IP
Telephone. The Avaya IP Telephone configuration is supported on the Avaya IP
Telephone with release 1.7 or later.
The IP Softphone is a TAPI-compliant application that functions over the IP network
in conjunction with Avaya IP Solutions. The Avaya IP Softphone is connected either
directly using a network interface card (NIC), by a dial-up modem with a point-topoint (PPP) account, or by a cable/DSL/ISDN connection. The Avaya IP Softphone
Release 5.1 will run on Microsoft Windows 2000 and Windows XP software. Avaya
IP Softphones include our Remote Help Desk Support offer, which can help to
simplify installation and ease deployment.
Key Features
Access to Avaya Communication Manager Station Features
The IP Softphone accesses the same features as those available on your office
phone. You have two easy-to-use graphical user interfaces—a call bar view and a
telephone image view. The call bar view shows you the basic Windows user
interface. The telephone image view provides you with an on-screen picture of the
telephone administered for your extension.
Local Phone Directory
You can create a local phone directory on your PC or laptop for your IP Softphone.
Place calls from the directory by selecting a contact name and clicking on the
telephone icon, or by hitting the Enter button on your keyboard.
Call Log and Redial List
Track incoming and outgoing calls with the Call Log. You can dial from the call log
and also dial the last number that was called by using the Redial button.
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Interface with TAPI-Compliant Personal Information Managers
You can configure TAPI-compliant Personal Information Managers (PIMs), giving
you direct-dial access from the PIM contact list. The Softphone will then take
control of that call to provide the appropriate station features.
Instant Messaging (IM)-only mode of operation
Users can invoke the Instant Messaging (IM) features of Softphone independently
of other functions without logging into Avaya Communication Manager. The
capability to support IM requires an Avaya Converged Communications Server.
Lotus Notes Integration
Users can lookup and dial from a number in a Lotus Notes directory and dial it.
Audio enhancements
Ringing calls can use an alternate sound card, allowing Softphone users to hear
ringing while on another call.
Bluetooth Headset Support
Bluetooth-compliant
wireless
headsets
implementations of IP Softphone.
are
supported
in
Windows
XP
Buddy-style Contact List
Contact lists resemble the buddy lists of users’ favorite IM applications.
Polycom ViaVideo II Integration
Avaya Integrator for Polycom video is also available (free of charge). This software
module for Avaya IP Softphone 5.1 enables an IP Softphone user to launch a video
call to another Softphone user with a mouse click or button press if each are
equipped with a Polycom ViaVideo II personal videoconferencing system. Robust
features of Avaya Communication Manager such as conference, transfer and hold,
can be applied to the video call with no reconfiguration requirement. This
integration takes Avaya and Polycom one step closer to making video as easy as
voice.
Both IP Softphone R5.1 and Avaya Integrator for Polycom Video applications are
available on the www.support.avaya.com website.
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Additional Features
¾
Multiple call appearances and one-button access to frequently used features,
such as Answer, Conference, Transfer, Hold, Mute, Redial, and Volume
Control
¾
Message waiting indicator, to signal new voice mail messages
¾
Speed dial, for one-click dialing
¾
Lightweight Directory Access Protocol client, giving you access to employee
or contact addresses and telephone numbers
¾
Access to the Avaya Communication Manager Directory, for telephone
numbers on the call processor directory
¾
Incoming Call Alerter (audio and visual)
¾
Online Help
¾
Multiple language support
¾
Survivability against Denial of Service Attacks
¾
Password protected login sessions
¾
G.711, G.729a, G.723.1a audio voice CODECs
The PC Requirements for supporting Avaya IP Softphone are as follows:
¾
IBM PC or compatible PC with the Intel Pentium III 300 MHz (400 MHz
recommended for Road Warrior) or compatible processor
¾
Hard disk with at least 50 MB of space available
¾
RAM Requirements:
Operating System Road Warrior
Telecommuter,
Shared Control
Windows XP
256 MB
128 MB
Windows 2000
128 MB
64 MB
¾
For the option of ringing on the PC, a sound device.
¾
For Road Warrior configuration only, a sound device that supports full-duplex
operation (both parties can talk and hear each other at the same time)
¾
For Road Warrior configuration only, a speaker/headset, and a microphone
¾
For Telecommuter configuration only, an available telephone line
¾
Network Interface Card (NIC) for local area network (LAN) connectivity
and/or a modem (28.8 Kbps or faster) for dial-up networking
¾
CD-ROM drive (if installing from CD)
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¾
Microsoft Internet Explorer 5.5 or higher to view the online help and to
support the ability to dial numbers from web pages
¾
One of the following operating systems:
¾
Microsoft Windows XP Home or Professional with Service Pack 1 or higher
¾
Microsoft Windows 2000 Professional with Service Pack 3 or higher
Additional Notes:
1. The minimum RAM requirements assume that IP Softphone is the only
application running on the PC. In reality, most PCs will be running one or more
other applications concurrently with IP Softphone. If you have not yet purchased
a PC on which to run IP Softphone, you should select a PC that exceeds the
minimum requirements. How far your PC should exceed these requirements
depends on how memory intensive the other applications are that you typically
use. If you frequently use multiple applications concurrently that are memory
intensive (such as a web browser), you should select a PC that has a fast
processor (that is, 800 MHz or faster) and 128 MB or more of RAM.
2. Avaya IP Softphone will not work if Internet Connection Firewall (ICF) is enabled
on the Windows XP machine.
3. Windows 95, Windows 98, Windows Millennium Edition, Windows NT, Windows
2000 Advanced Server, Windows 2000 Datacenter Server, Windows XP Server,
Windows XP Advanced Server, Windows XP Datacenter Server, IBM OS/2, Apple
MAC OS, Linux and UNIX are NOT supported. Any operating system that is not
listed in the PC requirements above is also NOT supported.
Touch-screen systems have not been certified to work with Avaya IP Softphone.
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 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.
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Part 1 – Section 4 - Voice Terminal Instruments
Avaya Response:
Comply. IP Softphone makes it easy to place and receive phone calls from your PC
or laptop, making it ideal for teleworking applications. It gives the user a choice of
simple Graphical User Interfaces (GUIs) and integrates well with Microsoft desktop
applications such as Outlook and Internet Explorer, enabling click to dial and screen
pops. There are options to add Instant Messaging and Presence as well as
videoconferencing for enhanced collaboration capabilities.
Product Details
Avaya IP Softphone integrates a flexible IP telephone client with a SIP/SIMPLEbased Instant Messaging (IM) client. It incorporates a contact list of other IP
Softphone users and makes both phone and IM presence visible to other users. It is
simple to toggle between the softphone and IM applications. The IM and presence
capabilities require registering with the Avaya SIP Enablement Service (SES)
platform, which is available separately.
The IP Softphone telephone client can be configured for VoIP mode or Multi-phone
mode. The VoIP configuration is suitable for users who have a single, broadband
connection from their home office or remote location. In VoIP mode, the audio and
call control signaling are run across the IP network for a "pure voice over IP"
configuration, and offers a great amount of flexibility due to the ubiquity of IP
networking. The Multi-phone configuration is ideally suited for users working from a
remote office with two lines for remote access. With this option, call control
signaling is maintained and delivered across the IP network, but the voice is
delivered across a second line to either a public switched telephone network or
digital line to help ensure toll-quality voice. This capability can be extended to any
touch-tone phone or cellular phone. In the Multi-phone configuration, the Avaya
communications server "binds" the two connections as a single session.
The IP Softphone can also be configured for shared control of either Avaya IP
telephones or Avaya digital telephones. The Avaya IP telephone configuration
enables users to log into and control their Avaya IP telephone from the Avaya IP
Softphone. Users can speak and listen through their telephone, but unlike the MultiPhone configuration, users can make and handle calls from both the Avaya IP
Softphone interface and the IP telephone. This feature can help improve
productivity by integrating the IP Softphone capabilities and Personal Information
Managers, such as Microsoft Outlook, with the IP Telephone.
IP Softphone offers two primary user interfaces; a simple to use "call bar" view in
which features are accessed via menu options and a "picture of phone" view that
emulates the user's desktop telephone.
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Part 1 – Section 4 - Voice Terminal Instruments
With the introduction of Avaya IP Softphone 5.1, there is a new mode of operation
for the IP Softphone called IM-only mode. In this mode, the Instant Messaging
capability of IP Softphone can be used independently of the other modes of
operation. For those who like to use IP Softphone in telecommuter mode or road
warrior mode but not shared control mode, IM-only mode enables them to use IM
from their desktop when they are "in the office". (Note that Instant Messaging does
require the Avaya SIP Enablement Services or SES Platform in all instances).
Please note that the user's phone presence (on-hook, off-hook, etc) can be visible
to other presence-enabled IP Softphone users in all but the IM-only mode of
operation. Of course, this requires registration with the Avaya SES Platform
An integrator module is available for Avaya IP Softphone that streamlines video call
setup using standard web cams. It also helps integrate desktop video conferencing
with Polycom group videoconferencing systems. Known as Avaya Video Integrator
2.0, this software module automatically identifies whether the far-end is videoenabled and establishes a video connection automatically. Avaya IP Telephony
features such as hold, transfer, forward and cover can be applied to a video call.
When desired - the video call can be dropped and the audio path remains active.
Avaya Video Integrator 2.0 will work in shared control mode, telecommuter mode
or road warrior mode. When used in shared-control mode the Avaya IP or Digital
telephone can be used for the audio path. Avaya Video Integrator 2.0 requires IP
Softphone 5.2 and is available as a free download from the Avaya Support web site.
An illustration of the Avaya IP Softphone is included in Appendix 3, PowerPoint
Illustrations.
Key Features
Benefits
Integration of Instant Messaging (IM)
and Presence with IP Softphone
IP Softphone users can send an Instant
Message to each other in addition to using
the phone application, as well as track each
other's phone or IM presence. A contact list
of 50 to 100 users with presence availability
can be maintained by each user.
Integration of IP Softphone and desktop
video
Avaya Video Integrator enables you to
automatically establish a video call when
your desktop and the far-end are video
equipped.
December 1, 2006
Users can collaborate more effectively via
their channels of preference.
Streamlines the setup of desktop video
calls which saves time and effort.
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Part 1 – Section 4 - Voice Terminal Instruments
Key Features
Benefits
Dialing from MS Outlook, Lotus Notes,
LDAP Directories, Web pages and
Windows applications
IP Softphone enables you to click and dial
from your Microsoft Outlook contact list,
your Lotus Notes contact list, an LDAP
directory, Web page or any Windows
application.
Helps improve user productivity and
efficiency by allowing users to spend less
time dialing with fewer dialing mistakes.
Improved Audio Support
Avaya IP Softphone now works with
Bluetooth wireless headsets and supports
ringing to an alternate sound card (so IP
Softphone users using a headset can hear
ringing while on a call).
Remote Access to Avaya Communication
Manager Station Features
The IP Softphone accesses the same
features as those available on the office
phone. There are two easy-to-use graphical
user interfaces: a call bar view and a
telephone image view. The call bar view
shows the basic Windows user interface. The
telephone image view provides an on-screen
picture of the telephone administered for the
user’s extension.
Better usability of IP Softphone
application.
With two easy-to-use graphical user
interfaces, preferences can be met and
efficiencies gained.
Local Phone Directory
A local phone directory can be created on
the PC or laptop for dialing from IP
Softphone. Place calls from the directory by
selecting a contact name and clicking on the
telephone icon, or by hitting the Enter button
on the keyboard.
Provides greater efficiency through fast
look up and easy dialing.
Call Log and Redial List
Track incoming and outgoing calls with the
Call Log. Dialing can be initiated from the
call log and also dial the last number that
was called by using the Redial button.
December 1, 2006
Track incoming and outgoing calls with the
Call Log.
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Part 1 – Section 4 - Voice Terminal Instruments
Key Features
Benefits
Shared Control of Avaya IP and DCP
Telephones
The option to use an IP or DCP telephone
handset for audio path while dialing from IP
Softphone is available.
Use IP Softphone application for directory
dialing on IP or DCP telephone.
Support of Multiple Languages
IP Softphone now supports many more
languages than English. Latin American
Spanish, Parisian French, Italian, German,
Brazilian Portuguese, Simplified Chinese,
Japanese, Russian, Korean are all supported.
Multiple call appearances and onebutton access
Multiple call appearances and one-button
access to frequently used features, such as
Answer, Conference, Transfer, Hold, Mute,
Redial, and Volume Control.
IP Softphone can be used more effectively
in more regions across the globe.
Improves productivity while away from
the office.
AES-based Media Encryption (AES)
The voice stream of an IP Softphone call can
be encrypted using the AES encryption
standard.
AES is a Federal Information Processing
Standard (FIPS) that specifies a
cryptographic algorithm for use by U.S.
Government organizations to protect
sensitive, unclassified information. Avaya
support of this standard in IP Softphone
helps ensure that sensitive voice
conversations remain private.
Message Waiting Indicator
Visual signal of the presence of new voice
mail messages.
Visual indicator of pending messages can
save time and increase efficiency of end
users.
Speed dial
Provides pre-programmed dialing capability
December 1, 2006
For one-click dialing.
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Part 1 – Section 4 - Voice Terminal Instruments
Key Features
Benefits
Lightweight Directory Access Protocol
client (LDAP)
Integration with the corporate directory
through LDAP
Access to the Avaya Communication
Manager Directory
For easy look up of co-worker access
information
Gives access to employee or contact
addresses and telephone numbers from
the corporate directory.
Helps access telephone numbers in Avaya
Communication Manager directory.
Incoming Call Alerter (audio and visual)
Both an audio and visual signal are given via
the IP Softphone upon receipt of an incoming
call.
Enables IP Softphone users to be more
efficient and responsive.
Outlook Integrator
Pop-up screen of Microsoft Outlook Contact
detail when caller ID matches Contact detail
phone number.
Interface with TAPI–Compliant Personal
Information Managers
The IP Softphone can be configured with
TAPI-compliant Personal Information
Managers (PIMs), giving direct-dial access
from the PIM contact list. The Softphone will
then take control of that call to provide the
appropriate station features list.
Additional functionality & security
capabilities built in
Password protected login sessions
Survivability against Denial of Service
Attacks.
December 1, 2006
Additional information regarding caller can
improve communication efficiency.
Direct-dial access from the PIM contact
list.
Helps provide a tightly secure solution for
remote workers.
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Part 1 – Section 4 - Voice Terminal Instruments
4.3.1.1 Teleworker Station User Application
The proposed PC client softphone solution may also be used by some station users as a
teleworker voice terminal outside the VoiceCon facility environment.
Vendor Response Requirement
Confirm that the proposed softphone solution can be used as a teleworker voice terminal
option. Indicate in your response if any optional hardware/software requirements are
required to support teleworker mode operations for deployment in a home, hotel, or
office environment.
Avaya Response:
Comply
Hardware Requirements
PC Requirements
¾
Intel Pentium 300 MHz (400 MHz recommended for Road Warrior) or
compatible processor Minimum of 30 Mb of available hard disk space
¾
A full duplex sound device (both parties can talk and hear each other at the
same time), speaker/headset, and a microphone (Road Warrior only).
¾
Network Interface Card for local area network connectivity and/or a modem
(28.8 Kbps or faster) for dial-up networking
¾
A CD-ROM drive, Microsoft Windows software-compatible VGA (or better)
adapter and pointing device (usually a mouse)
Operating System:
Microsoft Windows 98
¾
Telecommuter/, IP Telephone, Configuration: 32 Mb RAM
¾
Road Warrior, Configuration: 64 Mb RAM
Operating System: Microsoft Windows Me
¾
Telecommuter/, IP Telephone, Configuration: 32 Mb RAM
¾
Road Warrior, Configuration: 64 Mb RAM
Operating System: Microsoft Windows NT
¾
Telecommuter/, IP Telephone, Configuration: 64 Mb RAM
¾
Road Warrior, Configuration: 128 Mb RAM
¾
Operating System: Microsoft Windows 2000
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Telecommuter/, IP or Digital Telephone, Configuration: 64 Mb RAM
¾
Road Warrior, Configuration: 128 Mb RAM
¾
Operating System: Microsoft Windows XP
Telecommuter/, IP or Digital Telephone, Configuration: 128 Mb RAM
¾
Road Warrior, Configuration: 128 Mb RAM.
Software Requirements
Operating Systems
IP Softphone Release 5.X requires one of the following operating systems:
¾
Microsoft Windows XP (Professional or Home Edition)
¾
Microsoft Windows 2000 Professional with Service Pack 2 or later
¾
Microsoft Internet Explorer 5.5 or later to view the online help.
(Microsoft Windows 95, Windows NT Server, Windows 2000 Server and Datacenter
Server are not supported).
4.3.2 Soft Attendant Console
Attendant operator console requirements are to be satisfied using a PC client softphone
application. The attendant console application should include several distinct display
fields, such as: incoming call queue and active caller information; release loop keys;
feature/function keys; direct station selection (contact directory)/ busy lamp field; trunk
groups; minor/major alarms; and messaging. GUI capabilities must support drag & click
operations.
At minimum the following information and data must be available in the softphone screen
display: # 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.
December 1, 2006
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Part 1 – Section 4 - Voice Terminal Instruments
Avaya Response:
Comply. Avaya SoftConsole is a software attendant console that builds on the
features of the popular Avaya 302 Attendant Console. With its ability to search
internal and external directories, and display detailed caller information on up to six
calls simultaneously, Avaya SoftConsole brings more productivity to the attendant’s
experience. Avaya SoftConsole improves the user experience through a new
interface, comprehensive setup wizards, e-mail integration, and enhanced directory
capabilities.
Choice of two IP connections or DCP connection
o Voice over IP configuration (telecommuter)
o Dual connection (road warrior) for toll quality
¾ DCP connection using Avaya CallMaster VI
¾ Integrated iClarity for IP audio
¾ Directory lookup and dialing
¾ Integrated with directory management to support up to 100 directory
databases,
¾ Permanent and per call notes
The following are additional features of the Avaya Softconsole:
¾ Busy Lamp Fields (BLF), directory and display windows may all be on the
same screen and the same time.
¾ Flexible screen arrangement for the attendant that is saved from session to
session.
¾ Application window scales intelligently from a minimum useful size to full
screen. Useful information is added to the display as the attendant increases
the window size.
¾ On request line status such as on and off-hook displayed for the selected
entry in the directory window.
¾ Queue status display
¾ Feature buttons offered as tools in multiple tool bars with pop-up, full word,
tool tip displays for each.
¾ 32-bit Application
¾ Maximum of 100 directories
¾ Ability to generate e-mail to users at the click of a tool bar button or
keyboard command
¾ Step-by-step wizard for both installation and initial administration with help
and warning text presented with each step.
¾ Targeted to reduce service call volumes for installation assistance
¾
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Part 1 – Section 4 - Voice Terminal Instruments
Directory
The Master Directory Data Manager application is included as part of Avaya
Softconsole. This database application is specifically designed for directory data
management. It serves as information management tool–importing and
consolidating directory information from voice and data systems, and exporting it to
directory-enabled applications. MasterDirectory can import/export and transfer data
via standards-based protocols, including the following:
¾
ODBC – Open Data Base Connectivity
¾
LDAP – Lightweight Directory Access Protocol
¾
FTP – File Transfer Protocol
¾
SMTP – Simple Mail Transfer Protocol
¾
CSV – text delimited files
Using these protocols MasterDirectory can:
¾
Extract data from multiple sources
¾
Apply filters and business logic to consolidate data
¾
Populate directory services and databases for use by applications
For example, MasterDirectory can collect information from multiple Avaya Media
Servers, consolidate the data with Human Resource databases, and send the
processed data to an LDAP directory service used by phone attendant applications,
Internet white and yellow pages, and other applications.
As part of a group of applications called Avaya Integrated Management, the
optional Directory Enabled Management application erases the gap that existed
between telecommunications and messaging servers and other information sources,
such as your corporate databases. Now a change in one directory can trigger
changes across your organization, so you can create, update and access voice and
data information quickly and efficiently based on customer created rules and
policies or integration with third party applications.
Avaya Directory Enabled Management
Avaya Directory Enabled Management is a Windows client/server software solution,
which provides real-time Directory based Lightweight Directory Access Protocol
(LDAP) read/write access to servers running Avaya Communication Manager. Avaya
Directory Enabled Management provides the capability to keep this data (e.g.
station and subscriber data) synchronized with its image in the LDAP data store and
a rules engine that facilitates the management of these Avaya servers/applications
based on events (add/delete/modify) that take place at servers or applications. This
is a component-based centrally administered synchronization system that is highly
scalable and distributable.
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Part 1 – Section 4 - Voice Terminal Instruments
The Directory services supported include:
4.4
o
Netscape Directory Server 4.12
o
Microsoft Active Directory
o
IBM Directory Server 5.1
o
Novell NDS eDirectory 8.x
o
Sun™ ONE directory server 5.1
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.
Avaya Response:
Comply. We have proposed the Avaya 4690 Conferencing Station. The unit is
available in two models, one with extended microphones for large conference
rooms, and an office model, without microphones for smaller conference rooms.
The proposed solution does not include extended microphones. A picture of the
Avaya 4690 Conferencing Station is included in Appendix 3, PowerPoint
Illustrations.
Some benefits of the 4690 include:
¾
Access to complete Avaya Communication Manager feature set
¾
Emulation of a 4612 IP Telephone to Avaya Communication Manager
¾
Global ready (Icon labels & Class B EMC)
¾
Backlit, 4 Line text-based display (Eurofont only)
¾
4 fixed keys: On/Off hook, Redial, Mute, Volume Up/Down
¾
3 context sensitive soft keys below display
¾
Local phone features via 4 menu keys
¾
Power supply included
December 1, 2006
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Part 1 – Section 4 - Voice Terminal Instruments
4.8
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 applicable), and any other information you deem vital. Include an
illustration/photograph (PPT format, only) for each of these additional models.
Avaya Response:
Following is a brief description of the 4602, 4610, 4621, 4622 4625, and the EU24
Expansion Module, all of which are included in Appendix 3, PowerPoint Illustrations.
Avaya 4602 IP Telephone
The 4602 IP Telephone is an entry-level telephone with two programmable call
appearance/feature keys, ten fixed feature buttons, and display. The 4602SW offers
the same functionality plus an integrated two-port Ethernet switch. Both the 4602
and 4602SW can run either the H.323 protocol, for integration with traditional
Avaya IP Telephony Servers and Gateways, or the SIP protocol, for support of SIP
Communications Servers such as the Avaya Converged Communications Server as
well as 3rd party SIP proxies.
The 4602 includes the following features:
•
2 call appearances with LEDs
•
•
Fixed button with LED for voice mail
•
retrieval
•
Ten fixed feature buttons, including:
o Hold
o Transfer
o Conference
o Drop
Ethernet
network
10/100Base-T
connection with RJ-45 interface
Message waiting light (LED)
•
Supports G.711, G.729A, and G.729B
audio voice codecs
•
Supports H.323 V2
•
Supports 802.3af standard power over
Ethernet
o Redial
Avaya 4610SW IP Telephone
The 4610SW IP Telephone provides a medium screen graphic display, paperless
button labels, call log, speed dial, 12 programmable feature keys, Web browser,
and full duplex speakerphone. It also includes a two-port Ethernet switch. The
4610SW supports Unicode with R2.1 firmware.
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The 4610 includes the following features:
¾
12 Programmable Feature Keys
¾
Automatically labeled from the
system (no paper labels)
¾
10 Fixed Feature Keys: Hold,
Transfer, Conference, Drop,
Redial, Speaker, Mute, Drop,
Hold, and Volume Up & Down
¾
Mid-sized Graphical gray-scale
display (168 x 84 dots)
¾
3 Application buttons, along
bottom of the display
¾
Multiple language support built-in
(English, Dutch, Portuguese,
German, French, Italian, Spanish,
& KataKana)
¾
G.711, G.729A/B Voice Coders
¾
QoS Options of UDP Port
selection, Diffserv, 802.1p/q
¾
Support for Simple Network
management Protocol (SNMP)
version 2
¾
DHCP client and Statically
(Manual) Configurable IP
Addressing
¾
Multiple power options
¾
Speed Dial, Call Log, Web Browser
¾
Integrated Switched ports for
connection of PC
¾
Auto-negotiation provided
separately for each port
¾
10/100 Base T Ethernet
connections.
¾
Full Duplex Ethernet connectivity
¾
¾
802.3 Flow Control on full duplex
ports
Wall Mountable with included
desk/wall mount stand
¾
Message Waiting Indicator
¾
Supports VLAN
¾
¾
Full Duplex Speakerphone with
acoustic cavity for improved
sound quality
Downloadable Software for future
upgrade capability
¾
Hearing Aid Compatibility
¾
Icon button labelling with English
printing on the housing
¾
8 Personalized Ring Patterns
¾
Voice Media encryption
¾
3 position adjustable Desk Stand
¾
Integrated Headset Jack
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Part 1 – Section 4 - Voice Terminal Instruments
Avaya 4621 IP Telephone
The 4621SW IP Telephone is based on the 4620SW with the same housing, feature
buttons, and software along with an enhanced backlit display. The 4621SW also
supports the new EU24BL (backlit expansion module) and also includes a 2 port
Ethernet switch. A new phone, the 4621SW will ultimately replace the 4620SW.
Benefits
¾
Improved Productivity with paperless labels, call log, speed dial, and web
browser features
¾
Simplified wiring connects to your IP network with 10/100 BaseT Ethernet
LAN connection
¾
Multiple power options, including support for power over Ethernet LAN
¾
Investment Protection with easy upgrades via downloadable software and
firmware
¾
Secure voice communication enabled by media encryption capabilities
Key Features
¾
24 Programmable Feature Keys
¾
Automatically labeled from the
system (no paper labels)
¾
Infrared port to support Personal Digital
Assistant (PDA) dialling and additional
future applications
10
Fixed
Feature
Keys: ¾ Multiple language support built-in (English,
Speaker, Mute, Drop, Hold, and
Portuguese, Dutch, German, Japanese,
Volume Up & Down
Russian, French, Italian, Spanish, &
Chinese)
¾ Large
Graphical
gray-scale
display
(168 x 132 dots)
¾
¾
3 Fixed Feature Keys below the ¾ G.711, G.729A/B Voice Coders
Display: Conference, Transfer,
Redial
¾
3 Application buttons, along ¾ QoS Options of UDP Port selection,
bottom of the display : Speed
Diffserv, 802.1p/q
Dial, Call Log, Web Browser
¾ Support for Simple Network management
Protocol (SNMP) version 2
¾
Integrated Switched ports for ¾ DHCP client and Statically
connection of PC
Configurable IP Addressing
¾
December 1, 2006
(Manual)
Multiple power options
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Part 1 – Section 4 - Voice Terminal Instruments
Key Features
Auto-negotiation
provided ¾ 10/100 Base T Ethernet connections.
separately for each port
¾ Wall Mountable with included desk/wall
¾ Full
Duplex
Ethernet
mount stand.
connectivity
¾ Message Waiting Indicator
¾ 802.3 Flow Control on full
¾ Downloadable Software for future upgrade
duplex ports
capability
¾ Supports VLAN
¾
¾
Full Duplex Speakerphone with ¾ Hearing Aid Compatibility
acoustic cavity for improved
sound quality
¾
Feature Key
interface jack
¾
For use with the EU24,
button expansion module
¾
7 position
Stand
¾
Integrated Headset Jack
December 1, 2006
Module
adjustable
(FKM) ¾ Icon button labelling with English printing
on the housing
24 ¾ 8 Personalized Ring Patterns
Desk ¾ Voice media encryption using Advanced
Encryption Standard (AES)
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Part 1 – Section 4 - Voice Terminal Instruments
Avaya 4622 IP Telephone
The 4622SW IP Telephone is built specifically for the contact center environment.
Based on the 4621SW, the 4622SW offers the same backlit display and array of
feature buttons.
The 4622SW provides two headset jacks and eliminates
components not required within the call center including a handset and speaker.
Key Features
¾ 24 Programmable Feature
Keys
¾ Full Duplex Speakerphone
with acoustic cavity for
improved sound quality
¾ 10 Fixed Feature Keys:
Speaker, Mute, Drop, Hold,
and Volume Up & Down
¾ Large Graphical gray-scale
display
(168 x 132 dots)
¾
¾
3 Fixed Feature Keys below
the Display: Conference,
Transfer, Redial
3 Application buttons, along
bottom of the display :
Speed Dial, Call Log, Web
Browser
December 1, 2006
¾
Infrared port to support Personal Digital
Assistant (PDA) dialing and additional
future applications
¾
Multiple language support built-in
(English, Portuguese, Dutch, German,
Japanese, Russian, French, Italian,
Spanish, & Chinese)
¾
G.711, G.729A/B Voice Coders
¾
Multiple power options
¾
Integrated Headset Jack
¾
Integrated Switched ports for connection
of PC
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Part 1 – Section 4 - Voice Terminal Instruments
Avaya 4625 IP Telephone
The 4625 IP Telephone provides a brilliant color display. Also based on the 4621 in
terms of line appearances and feature buttons, the 4625 is an ideal device to
support applications within the Avaya Phone Application Suite.
Key Features
Key Features
¾ 10 Fixed Feature Keys: Speaker, ¾ Large Graphical gray-scale display
Mute, Drop, Hold, and Volume Up &
(168 x 132 dots)
Down
¾ 24 Programmable Feature Keys
¾ Full Duplex Speakerphone with
acoustic cavity for improved sound
quality
¾ Infrared port to support Personal
Digital Assistant (PDA) dialing and
additional future applications
¾ Multiple language support built-in
¾ QoS Options of UDP Port selection,
(English, Portuguese, Dutch, German,
Diffserv, 802.1p/q
Japanese, Russian, French, Italian,
¾ Integrated
Switched
ports
for
Spanish, & Chinese)
connection of PC
¾ 3 Fixed Feature Keys below the ¾ 3 Application buttons, along bottom
of the display : Speed Dial, Call Log,
Display: Conference, Transfer, Redial
Web Browser
¾ Multiple power options
EU24/EU24BL Expansion Unit
The EU24/EU24BL Expansion Unit is an optional device that extends the number of
call appearances and Feature buttons available on the telephone. The EU24 and
EU24BL Expansion Units are identical in terms of features and functionality. The
EU24BL has a backlit display area and is used with different telephone models than
the EU24, which does not have a backlit display.
Use the EU24 with 4620/4620SW IP Telephone and the EU24BL with 4621SW IP
and 4622SW IP Telephone.
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Part 1 – Section 5 - Call Processing Features
5.0
Call Processing Features
The proposed communications system should have a robust list of call processing
features supporting station user, attendant, and system operations.
Avaya Response:
Comply. Avaya Communication Manager includes more than 700 features.
Descriptions of the most widely used features are included in Appendix 1, Avaya
Communication Manager Feature Descriptions.
5.1
Station User Features
It is required that the proposed communications system support the following list of
station user features. Definitions for most listed features may be found in PBX Systems
for IP Telephony (2002), written by Allan Sulkin and published by McGraw-Hill
Professional.
Table 9 Station User Features
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
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EMERGENCY ACCESS TO ATTENDANT
EXECUTIVE ACCESS OVERRIDE
EXECUTIVE BUSY OVERRIDE
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
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Part 1 – Section 5 - Call Processing Features
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 and/or software, e.g., CTI application server,
because they are not included as part of the standard generic software package.
Avaya Response:
We comply to all of the features listed above, as described in the book, PBX
Systems for IP Telephony (2002), written by Allan Sulkin and published by McGrawHill Professional, except to the STEP CALL feature. For this feature, we partially
comply, as explained below.
Step Call
SULKIN DEFINITION: Step Call allows a station user or attendant, after dialing a busy
station, to dial an idle station by simply dialing an additional digit. The feature can
be implemented only if the dialed digits of the first dialed number and the second
number are identical, except for the last digit.
Partially comply. While we support this functionality on an attendant basis, it is not
supported on an individual user basis. If the functionality is needed for designated
users in VoiceCon’s enterprise, those users can be given the function on an
individual basis.
It is important to note that all of the features listed above, as well as many others
listed in Appendix 1, Avaya Communication Manager Feature Description, are
standard and supported on the current release of the software. Appendix 2 contains
a list of Avaya Communications Manager Standard/Optional Features.
5.1.1 Additional Station User Features
Vendor Response Requirement
Provide a listing of proposed standard generic software station user features that are not
included in Table 9 that VoiceCon may find of use and benefit.
Avaya Response:
The Avaya Communication Manager has over 700 features. A listing of the features
is provided in Appendix 1, Avaya Communication Manager Feature Descriptions.
Appendix 2 contains a list of Avaya Communications Manager Standard/Optional
Features. A couple of the more beneficial features offered as standard but not listed
above are:
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Part 1 – Section 5 - Call Processing Features
Posted Messages
The Posted Messages feature allows users to provide callers with a displayable
message offering a reason they are not available. This helps organization members
work together more efficiently. The system provides 30 messages to choose from,
15 of which are fixed (system messages) and 15 administrable (custom messages).
There are 15 system messages (fixed) as listed below:
¾ In Meeting
¾ Out To Lunch
¾ Away From Desk
¾ Do Not Disturb
¾ Out All Day
¾ On Vacation
¾ Gone For The Day
¾ Out Sick
¾ On Business Trip
¾ With Client
¾ Working From Home
¾ On Leave
¾ Back In 5 Minutes
¾ Back In 30 Minutes
¾ Back In 1 Hour
Though Intercom calling is listed, Automatic Intercom is worth noting due to the
benefits they provide for Secretary to Executive Office arrangements.
Intercom — Automatic
The Automatic Intercom feature allows two users to talk together easily. Calling
users press the Automatic Intercom button and lift the handset. The called user
receives a unique intercom ring and the intercom lamp, if administered, flashes.
With this feature, users who frequently call each other can do so by pressing one
button instead of dialing an extension number.
Conference/transfer display prompts
Conference/transfer display prompts are based on the user’s class of restriction
(COR). The display prompts are based on the user’s COR, independent of the select
line appearance conferencing and no dial tone conferencing feature. The display
messages vary depending on the activation of the two features, but the choice of
displaying the additional information or not is dependent on the station user’s COR.
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Conference/transfer toggle/swap
The conference/transfer toggle/swap feature allows users to toggle between two
parties in the middle of setting up a conference call prior to connecting all parties
together, or to consult with both parties prior to transferring a call. The display also
toggles between the two parties.
No dial tone conferencing
This feature can eliminate user confusion over receiving dial tone when trying to
conference two existing calls. It skips the automatic line selection if there is already
a party on hold or an alerting line appearance. Help messages help guide the user.
This feature is assigned on a system wide basis.
No hold conference
This feature allows a user to automatically add another party to a conference call
while continuing the conversation of the existing call. The new party is
automatically entered into the conversation as soon as the call is answered. An
optional tone can be provided prior to the party being added to the call. NOTE: The
calling station cannot hold, conference, or transfer an Emergency Access to
Attendant call. This applies to both the traditional means of using these features,
and to the no-hold method of using these features. After dialing is complete, if the
No Hold Conference is not answered within the time specified in an administered
“timeout” field, the No Hold Conference call is deactivated.
Select line appearance conferencing
If you are in a conversation on line “b” and another line is on hold or an incoming
call is alerting on line “a”, and then pressing the CONF button bridges the calls
together. Using the select line appearance feature on Communication Manager, the
user has the option of pressing a line appearance button to complete a conference
instead of pressing CONF a second time. This feature only applies if the line is
placed in soft hold by pressing the CONF button. This feature never applies if the
soft hold was due to pressing a TRANSFER button.
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Part 1 – Section 5 - Call Processing Features
Selective conference party display, drop, and mute
The selective conference party display, drop, and mute feature allows any user on a
digital station with display or on an attendant console to use the display to identify
all of the other parties on a two-party or conference call. The user would press a
feature button while on the call that puts the station or console into conference
display mode. The user then can scroll through the display of each party currently
on the call by repeatedly pressing the feature button. The display would show the
party’s number and name (when available). The user could then do either of the
following:
¾ The user can selectively drop the party currently shown on the display with a
single button push. This can be useful during conference calls when adding a
party that does not answer and the call goes to voice mail.
¾ The user can selectively mute the party currently shown on the display with a
single button push. This puts the selected party in “listen-only” mode.
This can be useful during conference calls when a party puts the conference call on
hold and everyone on the call is forced to listen to music-on-hold. The user can
mute that party so the conference call can continue without interruption. The muted
party can then rejoin the call by pressing the # key on their telephone. CAUTION:
Station users must be careful when scrolling through the displays when using the
selective conference party display feature. The station hyperactivity feature will
take the station out of service if the user repeatedly scrolls through the displays at
high enough rates. This causes the station to be reset and the user is dropped from
the call.
Selective conference mute
Selective conference mute allows a conference call participant, who has a display
station, to mute a noisy trunk line. Selective conference mute is also known as far
end mute. Examples of noisy trunk lines that might need to be muted during a
conference call are:
¾ cell phones
¾ phones that utilize the Music-On-Hold feature
¾ phones with no mute capabilities
Selective conference mute only applies to trunk lines on the conference call, and
not to stations. Only one trunk line on the conference call can be selectively muted
at a time. This enhanced conferencing feature can be activated from any display
station with a “conf-dsp” button and an “fe-mute” button. The selective conference
mute feature works with any conference established through Communication
Manager, either a traditional 3 or 6 party conference or a Meet-Me conference.
NOTE: This feature requires that the enhanced conferencing feature be set to Y on
the “system-parameters customer-options” screen.
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Part 1 – Section 5 - Call Processing Features
5.2
Attendant Operator Features
It is required that the proposed communications system support the following list of attendant
operator features. Definitions for most listed features may be found in PBX Systems for IP
Telephony (2002), written by Allan Sulkin and published by McGraw-Hill Professional.
Table 10 Attendant Operator Features
ATTENDANT 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
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).
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Part 1 – Section 5 - Call Processing Features
Avaya Response:
Comply. All attendant features listed above are supported by Avaya Communication
Manager, no additional hardware or software other than the proposed Avaya
Softconsole. Appendix 1, Avaya Communication Manager Feature Descriptions,
provides additional information.
The proposed Avaya Softconsole requires a PC for each attendant that is provided
by VoiceCon. The specifications are provided in item 4.3 of this proposal.
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.
Avaya Response:
Comply. All attendant features listed above are supported by Avaya Communication
Manager, no additional hardware or software are required other than the proposed
Avaya Softconsole and the client provided PC. Appendix 1, Avaya Communication
Manager Feature Descriptions, provides additional information. One feature that is
very beneficial is Attendant Vectoring.
Attendant Vectoring
Attendant Vectoring provides a highly flexible approach for managing incoming calls
to an attendant. For example, with current night service operation, calls redirected
from the attendant console to a night station can ring only at that station and will
not follow any coverage path.
With Attendant Vectoring, night service calls will follow the coverage path of the
night station. The coverage path could go to another station and eventually to a
voice mail system. The caller can then leave a message that can be retrieved and
acted upon.
This means you can program your after hour or special condition calls to route
anywhere throughout your system you would like providing added customer service
and after hour coverage.
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Part 1 – Section 5 - Call Processing Features
5.3
System Features
It is required that the proposed communications system support the following list of system
features. Definitions for most listed features may be found in PBX Systems for IP Telephony
(2002), written by Allan Sulkin and published by McGraw-Hill Professional.
Table 11 System Features
SYSTEM FEATURES
ACCOUNT CODES
ADMINISTERED CONNECTIONS
ANSWER DETECTION
AUTHORIZATION CODES
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
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Part 1 – Section 5 - Call Processing Features
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
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
Avaya Response:
Comply. All features listed above are supported as standard on the Avaya
Communications Manager, Release 3.1, which provides over 700 features, too
many to list below.
Additional detail on the features listed above is provided in Appendix 1, Avaya
Communication Manager Feature Descriptions. In addition, we included a Standard
and Optional Feature Guide in Appendix 2.
The following capacity numbers (requested above) apply:
¾ CLASSES OF RESTRICTION
96
¾ CLASSES OF SERVICE
16
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Request for Proposal for an IP Telephony System
Part 1 – Section 5 - Call Processing Features
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.
Avaya Response:
Comply. The Avaya Communication Manager has over 700 features. A listing of the
features is provided in Appendix 1, Avaya Communication Manager Feature
Descriptions.
Appendix 2 contains a list of Avaya Communications Manager
Standard/Optional Features. One quick feature to note is:
Holiday vectoring
With holiday vectoring, a flexible approach for managing incoming calls on special
dates is available. Holiday vectoring allows for branching and routing of calls based
on information about special schedules. The special schedules are recorded in
tables, each of which can hold up to 15 special dates or ranges of dates. Holiday
vectoring makes it possible for up to 10 tables to be treated differently in vector
processing.
5.4 Mobility Features
VoiceCon requires that the proposed IPTS support a variety of features and applications to
support its mobile workforce.
Comply. Avaya offers numerous Mobility solutions:
¾
Extension to Cellular
¾
IP Softphone
¾
WLAN Gateway/Wireless Telephones
¾
Mobile Convergence
¾
IP Softphone for Pocket PC
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Part 1 – Section 5 - Call Processing Features
Inside the Building
Outside the Building
Avaya Communication Manager
running on Avaya Media
Server/Media Gateway
PSTN
Wired
Network
Infrastructure
W310
WLAN
Gateway
Dual-network
Motorola Mobile
Office Device
(2H 04)
W110 WLAN
Access Points
Avaya Extension to Cellular
Avaya Speech Access to UCC
Avaya 3616/3626
IP Wireless
IP Softphone
for Pocket PC
IP Softphone
Extension to Cellular
Please refer to description in 5.4.2 below.
IP Softphone
Please refer to description in 4.3
Avaya W310 WLAN Gateway
We offer the Avaya W310 WLAN Gateway, a 16 port standards-based 802.3af
Power over Ethernet-enabled switch with 2 GbE uplinks. It provides centralized
intelligent management of the associated access points. Our wireless telephone sets
will also operate off of other standards based access points.
The Avaya W110 WLAN Access Point is simple, cost-effective and easy to deploy. It
is an integrated 802.a/b/g single radio device: operating in one mode at a time. It
accepts many of the external antennae already offered by Avaya to extend range if
necessary.
The Avaya W310 and W110 provide an excellent foundation to extend powerful
Avaya Communications Manager Productivity features to 802.11-enabled endpoints
such as Avaya 3616 and 3626 IP Wireless Telephones. Avaya IP Softphone client
software can be loaded on Windows-based laptops and PDA’s to provide enterpriseclass telephony to mobile employees. Finally, future solutions such as dual-mode,
802.11/cellular business phones can be supported as well: providing enterprises
with seamless converged communications across networks.
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Part 1 – Section 5 - Call Processing Features
Avaya IP Wireless Telephone Solutions
Avaya resells wireless telephone solutions from SpectraLink, who offers a complete
portfolio of wireless telephone systems for the workplace. SpectraLink is solely
dedicated to meeting the communication needs of mobile workers. To that end,
SpectraLink has developed products that integrate with all enterprise telephony
technologies, from the most widely used traditional PBX systems to the leadingedge IP-based convergence platforms. The systems can be designed with dedicated
wireless infrastructure for minimal administration and high reliability, or the
systems can utilize standards-based wireless LAN infrastructure to serve mobile
voice and data requirements.
No matter what technology an enterprise uses today, or plans to implement
tomorrow, SpectraLink has a wireless telephone product that delivers the highest
level of functionality along with excellent voice quality, durable handsets, high
reliability, and backed by the industry’s best service and support programs.
Avaya offers two IP Wireless Telephones – the Avaya 3616 IP Wireless Telephone
and the Avaya 3626 IP Wireless Telephone. The 3616 is a lightweight, executive
handset with a cell phone-like form factor, 4 hour talk time and 80 hour standby
time and a backlit 128 x 64 pixel display. It is designed for general enterprise use.
The 3626 uses a ruggedized form factor and has the same battery life and display
as the 3616. In addition, the 3626 adds a “push to talk” capability that enables
3626 telephones to act as walkie talkies over an 802.11b-compliant Wireless LAN.
It is designed for industrial and institutional applications.
Both the 3616 and 3626 phones are optimized for Avaya IP telephony and emulate
the wired 4606 IP Telephone. They require the same components as the existing
3606 IP Wireless Telephone, namely the Avaya Voice Priority Processor, and for
non-IP installations, the NetLink 150 Telephony Gateway.
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Avaya 3616 IP Wireless Telephone
Avaya 3626 IP Wireless Telephone
Seamless Converged Communications across Networks (SCCAN)
Please refer to description in 5.4.3 below.
Avaya IP Softphone for Pocket PC
The Avaya IP Softphone for Pocket PC enables workers to be mobile while accessing
comprehensive and powerful enterprise-class telephony on their Pocket PC based
devices.
With Avaya IP Softphone for Pocket PC, you can experience powerful telephony
functions from any location with the same feature functionality as if you were in
your office. In conjunction with the wireless capabilities of the Pocket PC, the Avaya
IP Softphone for Pocket PC gives you full mobile desktop capability via standard offthe-shelf Pocket PC devices and standard 802.11 interfaces. It leverages your
existing Avaya communications server and provides increased mobility options and
value—with no specialized hardware required.
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The Avaya IP Softphone for Pocket PC is a downloadable application for customers
who own an Avaya IP Softphone license. It delivers the full set of Avaya
Communication Manager call processing features via a graphical display of your
Avaya multi-line phone, with its identical extension number, speed dial buttons, and
personal feature settings. Mobile workers can receive calls virtually anywhere and
remote workers can connect to your enterprise with wireless local area networks
(LANs) and virtual private networks (VPNs). The Avaya IP Softphone for Pocket PC
is perfect for environments where communications are critical, and cellular phones
are not permitted—such as hospitals—and for other industries with large, wireless
enabled facilities—like manufacturers and universities.
Whether you're on the road, at the warehouse, in the clinic, or just in a meeting
room down the hall, the Avaya IP Softphone for Pocket PC helps you stay
connected, responsive, and productive.
Avaya IP Softphone for Pocket PC
Key Features of IP Softphone for Pocket PC
¾
Graphical User Interface - the application includes an easy-to-use
graphical user interface (GUI) mirroring your desk telephone. It provides
multiple call appearances and line status indicators, as well as conference,
transfer, hold, mute, and drop buttons. You have visual notification of
messages through the Message Indicator icon.
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¾
Access to Avaya Communication Manager Features – the application
supports access to the full set of features provided by Avaya Communication
Manager, such as incoming caller ID, Directory, and Bridged Call
Appearances. Avaya IP Softphone for Pocket PC simplifies communications by
integrating with other Pocket PC applications. For example, you can dial an
entry in your contact list with a single click, receive voice mail notification,
and toggle to e-mail and other applications.
¾
Connection Choices – choice of pure Voice-over-IP (VoIP) connection or
telecommuter mode—where call control is over IP but voice is over circuit
switched connection—for the highest quality audio.
¾
VPN Support – Virtual private network (VPN) support, providing secure
wireless connectivity. Remote workers can utilize the Avaya IP Softphone for
Pocket PC using your wireless network at home or a remote office. You also
have VPN access to your corporate network.
¾
Media Encryption – IP Softphone for Pocket PC includes an AES-based
media encryption option for users with stringent security requirements. In
addition, alternate gatekeepers are supported for better resiliency; and
shared control of DCP and IP Telephones. In addition, native control of
2402/2420 and 4602/4620 telephones is available.
¾
Network Diagnostics Tools – the application includes built-in tools for
trouble shooting the status of your network. A built-in "ping" allows you to
verify basic network connectivity to your Avaya communications server. You
also have access to the following information about how your Avaya IP
Softphone for Pocket PC is performing:
o
Currently active codec (e.g., G.711 Mu Law or G.711 A Law)
o
RTP packet size
o
IP address (and port) of the destination that is receiving your
generated VoIP packets
o
Local IP address (and port) of your PocketPC
o
Approximate incoming and outgoing bandwidth you are using for the
active call
o
Approximate incoming and outgoing jitter buffer delays as they are
monitored and adjusted automatically for maximum voice quality.
o
Real time messages informing you of network congestion resulting in
degradation of voice quality.
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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.
Avaya Response:
For VoiceCon employees who work out of the main office—on the road or at home—
the Avaya IP Softphone creates one simple, easy-access method for accessing the
office telephone. Once the Softphone software is installed on your PC or laptop, you
can place and receive calls—and access voice mail messages—using high-quality
iClarity IP Audio. You can also access the powerful Avaya Communications Manager
Station features such as multiple call appearances, caller ID, conference, as well as
speed dial, send all calls, and message waiting.
The Avaya IP Softphone allows users to easily make and receive calls using Voiceover-IP (VoIP), from any location using a simple graphical user interface on a PC or
laptop computer screen. Users can even select a graphical interface that looks just
like a desktop phone.
Avaya IP Softphones can be set up for one of two configurations:
¾
Road Warrior
¾
Telecommuter
Road Warrior Configuration
The Road Warrior configuration is suitable for users who may have only one line for
remote access, for example, someone working from a hotel room while traveling.
Using the Road Warrior mode, the voice or audio is delivered across the IP network
for a "pure voice over IP" configuration, and offers a great amount of flexibility due
to the ubiquity of IP networking.
Telecommuter Configuration
The Telecommuter configuration is ideally suited for users working from a remote
office with two lines for remote access. With this option, feature/access control and
signaling is maintained and delivered across the IP network, but the voice is
delivered across a second line to either a public switched telephone network or
digital line to help ensure toll-quality voice. This capability can be extended to a
cellular, PCS, or GSM phone. In the Telecommuter configuration, the Avaya
communications server "binds" the two connections as a single transaction or
session.
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Avaya IP Softphone Mobility
Avaya IP Softphone – Key Features
¾
Access to Avaya Communication Manager Station Features – the IP
Softphone accesses the same features as those available on your office
phone. You have two easy-to-use graphical user interfaces—a call bar view
and a telephone image view. The call bar view shows you the basic Windows
user interface. The telephone image view provides you with an on-screen
picture of the telephone administered for your extension.
¾
Local Phone Directory – You can create a local phone directory on your PC or
laptop for your IP Softphone. Place calls from the directory by selecting a
contact name and clicking on the telephone icon, or by hitting the Enter
button on your keyboard.
¾
Call Log and Redial List – Track incoming and outgoing calls with the Call
Log. You can dial from the call log and also dial the last number that was
called by using the Redial button.
¾
Interface with TAPI-Compliant Personal Information Managers – You can
configure TAPI-compliant Personal Information Managers (PIMs), giving you
direct-dial access from the PIM contact list. The Softphone will then take
control of that call to provide the appropriate station features.
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¾
Instant Messaging (IM)-only mode
Instant Messaging (IM) features
functions without logging into Avaya
to support IM requires the SIP
Communications Server.
of operation – Users can invoke the
of softphone independently of other
Communication Manager. The capability
capabilities of an Avaya Converged
¾
Lotus Notes Integration – Users can lookup and dial from a number in a
Lotus Notes directory and dial it.
¾
Audio enhancements – Ringing calls can use an alternate sound card,
allowing Softphone users to hear ringing while on another call.
¾
Bluetooth Headset Support – Bluetooth-compliant wireless headsets are
supported in Windows XP implementations of IP Softphone.
¾
Buddy-style Contact List – Contact lists resemble the buddy lists of users’
favorite IM applications.
¾
Enhanced desktop integration via click-to-dial from Internet Explorer page
and name look-ups and dialing from Microsoft Outlook Contact lists and LDAP
directories
¾
Improved user productivity via shared control of DCP telephones (
¾
Picture of phone support for 4620/4620SWAvailable
Communication Manager mobility package
as
part
of
Additional Features
¾
Multiple call appearances and one-button access to frequently used features,
such as Answer, Conference, Transfer, Hold, Mute, Redial, and Volume
Control
¾
Message waiting indicator, to signal new voice mail messages
¾
Speed dial, for one-click dialing
¾
Lightweight Directory Access Protocol client, giving you access to employee
or contact addresses and telephone numbers
¾
Access to the Avaya Communication Manager Directory, for telephone
numbers on the call processor directory
¾
Incoming Call Alerter (audio and visual)
¾
Online Help
¾
Multiple language support
¾
Survivability against Denial of Service Attacks
¾
Password protected login sessions
¾
G.711, G.729a, G.723.1a audio voice CODECs
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Part 1 – Section 5 - Call Processing Features
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:
Avaya Response:
Avaya one-X Mobile Edition for Symbian (Dual-Mode)
The Dual-Mode capability allows a user to roam calls between cellular and Wi-Fi
networks using SIP telephony. Moreover, unlike SCCAN and IMS technologies, this
dual-mode implementation does not require any additional infrastructure since all
the intelligence is built in-to the software. With four different selectable modes, the
Nokia® dual-mode phone will use Assisted Handover mechanism to handoff calls
between the two networks. Nokia E60, E61, and E70 are supported at this time.
Avaya one-X Mobile Edition is a family of client software for leading mobile
Smartphone platforms that transforms a user's mobile phone into their office desk
phone. Avaya one-X Mobile Edition allows mobile employees to easily access
powerful features of Avaya Communication Manager IP telephony software such as
multi-party conference calling, call transfer, call coverage, abbreviated dialing and
more. This ability to integrate mobile devices into business operations provides
opportunities to greatly improve employee productivity while controlling
communications costs. Avaya one-X Mobile Edition has been implemented initially
on the S60 platform, the world’s leading Smartphone platform.
Product Details
Avaya one-X Mobile Edition is a software client that enables easy access to the
features of Avaya Communication Manager while the user is mobile. Its initial
implementation is for Symbian OS-based mobile devices using the S60 platform. It
works in conjunction with the Extension to Cellular feature of Avaya Communication
Manager.
Avaya one-X Mobile Edition becomes the "business mode" of operation for the
mobile device. When one-X Mobile Edition is active, calls made from the mobile
device appear as if they were made from the employee's desktop business phone.
Conversely, calls made to the employee's business phone number can be answered
from the mobile device. A full suite of business telephone features such as transfer,
conference, call coverage, extension dialing, and others are available to the mobile
device user. The mobile employee has only one business phone number to manage,
one voice mailbox to check, and one-X Mobile Edition can be turned on and turned
off as needed.
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Part 1 – Section 5 - Call Processing Features
Avaya one-X Mobile Edition uses a familiar Graphical User Interface to provide easy
access to powerful, productivity-enhancing business phone features. The
combination of access to these features plus always-available accessibility greatly
increases mobile employee productivity.
IT Managers also benefit from the deployment of Avaya one-X Mobile Edition. In
particular, mobile phones can be better integrated into business operations in that
because the mobile device is now an extension of the business phone system,
mobile calls can be recorded if required and tracked through call detail records for
bill back or other purposes.
Avaya one-X Mobile Edition also helps businesses maintain better control over their
assets by providing a way for business numbers to remain with the enterprise upon
employee termination or resignation. Employees don't need to give their mobile
phone number as their primary contact; they can give their business number only.
Another financial advantage of one-X Mobile Edition is that operator charges can be
reduced for multinational firms by leveraging the corporate intranet for
international mobile to mobile calling.
Avaya/Nokia Dual Mode Solution Topology
¾
Nokia phone functions as off-PBX bridged extension when out of building
¾
Nokia phone functions as a SIP-based PBX extension when in building
¾
Nokia phone makes, receives, transfers calls through Avaya Communication
Manager and Avaya SIP Enablement Services platform
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Part 1 – Section 5 - Call Processing Features
Key Features
Benefits
Easy to user Graphical User Interface
Avaya one-X Mobile Edition utilizes a familiar S60 or S80
user interface with context-sensitive menu options.
One business number access
This feature is enables calls to the employee's business
number to be received by their mobile phone.
Location Transparency
Calls made from the mobile device using one-X Mobile
Edition appear to the called party as originating calling
party's business number, not mobile number.
Avaya one-X Mobile Edition
interface makes application easy
to learn and use.
Improved employee accessibility
which translates into better
customer service, improved
productivity, and faster decision
making.
Mobile number can be retained
by employee for personal use.
User control of Avaya one-X Mobile Edition
One-X Mobile Edition can be turned on and off as needed. Mobile employees can better
manage their work/life balance
by managing their availability to
accept business calls.
Dialing from Call logs and Contact lists
Avaya one-X Mobile Edition users can dial from their call
logs and local contact lists.
Improved user productivity and
ease of use.
Access to Avaya Communication Manager features
Avaya one-X Mobile Edition users have easy access to
These features greatly enhance
powerful enterprise telephony features like multi-party
the ability of mobile workers to
conferencing, multi-level transfer, extension dialing, mute, collaborate with colleagues,
hold and many others.
clients and others.
Single voice mailbox
By utilizing Avaya one-X Mobile Edition and the Extension
to Cellular function of Avaya Communication Manager, all
unanswered calls to the employee's business number go
to their business voice mail system; even those received
on the mobile phone.
International mobile call avoidance
Mobile to mobile calls can be very expensive when they
cross country borders. This capability enables enterprises
to utilize the enterprise telephony network for
international trunking of voice traffic.
December 1, 2006
©2006 Avaya Inc.
Enables the employee to be
more productive and efficient by
only checking one voice mailbox
for messages
Reduces international mobile-tomobile calling charges by
converting them to local mobileto-mobile calls.
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Avaya one-X™ Mobile Edition
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
¾
Simplified access to “Top 20” PBX features – a business softphone on a
mobile phone
o Full business call control: hold, conference, transfer, assistant
support, extension dialing
o It is your office phone!
One number access – incoming office phone calls extended to mobile
phone
One voice mail to check
Outgoing calls use corporate network
Centralized management & reporting
Easily switch between personal and business use of mobile phone
Support for 11 Languages
Supported on Nokia S60 and S80; other
platforms to follow
Enterprise dial plan support
Enable/Disable Extension to Cellular
Active appearance select
Enable/Disable Automatic call back
Call forwarding all
Call forwarding busy/no answer
Call forwarding deactivation
Call pick-up directed
Extension to Cellular
Conference on answer
Drop Last Added Party
Exclusion
Held appearance select
Idle appearance select
Last number dialed
Enable/Disable Send all calls
Transfer on hang up
Transfer to voice mail
Avaya Extension to Cellular
The Avaya Extension to Cellular capability offers one-number portability and onenumber access to any extension on a Communication Manager powered solution.
Inbound calling traffic no longer has to wait through numerous transfers only to
reach a voice mailbox. If the office is in a cellular-accessible area, Avaya Extension
to Cellular can give customers easy access to the people they want to reach,
helping increase customer satisfaction and raising productivity levels within the
organization.
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Part 1 – Section 5 - Call Processing Features
Avaya Extension to Cellular is a mobility solution that enables you to receive calls
where you are not where your desk is. Now you will get many of the productivity
capabilities you have on your desk phone right on your cellular set. Avaya
Communication Manager Capabilities such as Last Number Dialed, Call Forward,
Malicious Call Trace and Conference on Answer will help maximize the effectiveness
of employees, extend the capabilities of existing assets and enable you to respond
to ever changing conditions. Avaya Extension to Cellular is an efficient and cost
effective solution for reaching mobile workers anywhere in the world cellular
coverage is available. This includes the ability for students to use their cell phones
and room phones for the same switch capacity. Since it is operable with all major
cellular standards, such as TDMA, CDMA and GSM, you can install the solution
across your facilities—without having to think twice about what cellular carrier or
cellular standard is in use for a given area.
Avaya brings this solution to you through a software approach. By leveraging Avaya
Communication Manager, Avaya Extension to Cellular transparently bridges calls
received by an Avaya Communications Server, to any digital cell phone—regardless
of your location, Wireless Service Provider, and cellular standard your service
provider uses. Because your digital cell phone is bridged to your desk telephone
set, you can now be reached through one number—anytime, anywhere in the
world. Along with delivering complete mobility and one number availability, the
user will receive standard features for incoming calls, such as a caller ID—without
the requirement of a costly Wireless Office System (WOS).
With the Avaya Extension to Cellular solution, calls always remain at the specific
station dialed— whether the call is answered on a digital cell phone, your Avaya IP
telephone, multi-line desk set, or IP Softphone. So, whether you’re working from
your office, on your way to an important meeting or rushing home during your daily
commute, the Avaya Extension to Cellular will instantly deliver all your calls to your
digital cell phone. If you are not available, your calls will follow your desk phone's
coverage path—avoiding many, unanswered rings. With the Avaya Extension to
Cellular, important calls are not missed, and customers are satisfied because they
can reach the person they need to speak with.
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Part 1 – Section 5 - Call Processing Features
Capabilities built in to the Extension to Cellular feature include:
¾
Access to Avaya Communication Manager station features – Extension to
Cellular users can conference and transfer calls from a bridged cell phone as
well as pick up calls on two line appearances. There are twenty-four Avaya
Communication Manager features available via feature code access from a
bridged cell phone.
¾
Scalability – The maximum limit on the number of bridged cell phones is
bounded only by maximum station capacity of the Avaya Media Server/Avaya
Media Gateway combination.
¾
Feature status button – Extension to Cellular users can activate, deactivate,
and suspend Extension to Cellular service by using an administered Extension
to Cellular feature status button. The Extension to Cellular button remains lit
when service is enabled, off when service is disabled, and flashes at the
inverted wink rate when service is suspended through the optional timer. The
Extension to Cellular feature button is available on telephones that support
administrable feature buttons.
¾
Call Filtering – This feature helps ease customer concerns over recurring cell
phone expenses by limiting the calls extended to the cellular network for
Extension to Cellular users. Customers can choose to deliver, on a per-user
basis, only external calls (from a customer), only internal calls, all calls, or no
calls.
¾
Call classification – Extension to Cellular call filtering uses the same criteria
for classifying an external or internal call as the call coverage feature.
¾
Call Detail Recording – This feature enables logging of Extension to Cellular
calls and their details and creation of an export file that can be used by thirdparty Call Accounting Software systems for usage tracking.
For maximum flexibility, Avaya Extension to Cellular can be implemented using PRI,
BRI, IP and/or SIP trunks.
5.4.2.1 Shared directory number with IP desktop telephone instrument or PC client softphone for
inbound and outbound calls
Avaya Response:
Comply. See response 5.4.2 above.
5.4.2.2
Access to IPTS call answering and calling features (identify specific features
available to cellular handset user)
Avaya Response:
Comply. See response 5.4.2 above.
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Part 1 – Section 5 - Call Processing Features
5.4.2.3
Shared voice mailbox for desktop stations and cellular extensions
Avaya Response:
Comply. See response 5.4.2 above.
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
Avaya Response:
Comply. The optional Avaya Mobile for Series 60 Platform loads on a S60 2nd
edition mobile phone and provides an intuitive front-end to control Extension to
Cellular enterprise telephony features. Extension to Cellular treats a mobile phone
as if it were an extension of an office phone. Once enabled, Extension to Cellular
and Avaya Mobile allows one to answer calls, make calls as if they were placed from
their office number, enterprise transfer, enterprise conference and many other ACM
based features.
¾
Calls made from mobile phone appear as if from desktop
¾
Full suite of business telephony features (e.g., transfer, conference,
abbreviated dialing)
¾
Familiar GUI for easy access to phone features,
¾
single voice mailbox
¾
Improves business continuity for the enterprise
A picture of the Avaya S60 GUI interface is included in Appendix 3, PowerPoint
Illustrations.
5.4.2.5
Indicate if the cellular extension feature is proprietary to a specific cellular carrier
operator service
Avaya Response:
The Avaya Extension to Cellular offer is not specific to any cellular provider.
5.4.2.6
Identify any optional hardware/software required to support the cellular extension
feature if not a proposed IPTS generic software feature
Avaya Response:
The Avaya Extension to Cellular offer does not require specialized hardware. Five
free licenses come with each standard Avaya Communications Manager platform.
December 1, 2006
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Part 1 – Section 5 - Call Processing Features
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.
Avaya Response:
Avaya one-X Mobile Edition for Symbian (Dual-Mode)
In conjunction with Avaya Extension to Cellular Avaya offers this solution today,
please see response to question 5.4.2 for details
Planning Roadmap
Fall 2006/Winter 2007
¾
Avaya one-X Mobile Edition for S60 support for Nokia E-Series phones
(Single Mode) - done
¾
Avaya one-X Mobile Edition for S80 - done
¾
Avaya one-X Mobile Edition for Windows Mobile 5
¾
Avaya one-X Mobile Edition for S60 support for Nokia E-Series phones (Dual
Mode)
Later 2007
¾
Personal Communications Dashboard
¾
Directory, Call Logs, VVM, MWI, CLI, SMS pass-through
¾
Support for Blackberry devices
¾
Avaya one-X Mobile Edition for Windows Mobile 5 support for Dual Mode
December 1, 2006
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Part 1 – Section 6 - Systems Management
6.0.0 Systems Management
The proposed communications system must be administered, monitored, and
maintained through operations organized into five functional areas: Fault, Configuration,
Accounting, Performance and Security. All of the systems and devices in your proposed
solution should attempt to provide comprehensive operations in each area.
Operations for each area must be accessible through one interface regardless of the
underlying system or device being managed. If a proxy server is used for intermediate
operations, there must be at most one central database for each functional area.
Systems or devices may be accessed individually if no proxy server is used.
EXCEPTION: Optional call center solutions may provide its own set of FCAPS
management operations separate from the general enterprise communications solution.
Any supplied management applications must support decentralized access from any
distributed PC client across the HQ LAN/WAN infrastructure and remote dial-up PC
clients. It is also desirable for the applications to support a browser based user interface
for intensive remote operations.
Any supplied management applications may integrate information from the five
functional areas at the presentation level.
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.
Avaya Response:
Comply.
The proposed solution includes Standard Network Management. This powerful
management application will allow VoiceCon’s System Administrator to manage the
proposed Media Server, Media Gateways, Messaging Server, and optional Wireless
LAN Access Points from the same interface.
December 1, 2006
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Part 1 – Section 6 - Systems Management
Avaya Integrated Management Release 3.1
Avaya Integrated Management is a comprehensive set of tools that simplifies
management of converged network infrastructures. These applications are designed
to simplify system administration, provisioning, network management, and fault
and performance management operations. Avaya Integrated Management can help
customers improve their network uptime, increase staff productivity, and reduce
operating costs. The Avaya Integrated Management applications include the tools
that enable customers to:
¾
Configure, monitor, update, and optimize the performance of Avaya media
servers, gateways, and endpoints
¾
Monitor voice over IP traffic
¾
Manage Quality of Service (QoS) policies
¾
Control network quality
¾
Optimize performance
Avaya Integrated Management provides a complete solution for system
administration, network management, and provisioning of converged networks –
including both voice and data communications – while:
¾
Lowering management costs through simplified administration features
¾
Increasing productivity through a centralized management architecture
¾
Maximizing network reliability with network-wide monitoring and fault
management tools.
The solution helps customers achieve more efficient system management, less
system downtime, and a communications infrastructure that adds more value to an
organization than ever before.
Benefits
As a core component of the Avaya Communication Architecture, Integrated
Management encompasses the entire scope of Avaya offerings, including
Communication Manager software, media gateways and servers, Avaya DEFINITY®
Communications Servers, converged infrastructure switches, and a wide variety of
Avaya adjuncts such as Modular Messaging, INTUITY® Audix, Conversant, Call
Management System (CMS), MultiPoint Conferencing Unit (MCU), and Intuity
Interchange.
Integrated Management Application Components
The goal of Avaya Integrated Management is to provide a complete and
comprehensive set of management solutions. These applications are integrated,
such that an application can be launched from the Integrated Management
Homepage, from within the Avaya Network Management Console, or from a
customer-provided HP OpenView application. Additional Avaya Integrated
Management applications include:
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¾
¾
¾
Application Administration
o
Avaya Site Administration (ASA)
o
Avaya Voice Announcement Manager (VAM)
o
Avaya MultiSite Administration (MSA)
o
Integrated Management Database
Alarming and Monitoring
o
Avaya VoIP Monitoring Manager (VMM)
o
Avaya SMON Manager
o
Converged Network Analyzer
o
Avaya Fault and Performance Manager (FPM)
o
Proxy Agent
o
Network Management System Integration (NMSI)
Provisioning and Network Management
o
Avaya Network Management Console (NMC) with VoIP System View
o
Avaya Provisioning and Installation Manager (PIM)
o
Avaya Secure Access Administration
o
Avaya Software Update Manager
o
Avaya Network Configuration Manager
o
Avaya QoS Manager
o
Avaya VLAN Manager
o
Avaya Address Manager
o
Avaya Device Managers
Communication Manager Feature Administration
Specific information on the 700+ features of Avaya Communication Manager can be
found in specific Communication Manager documentation such as the Administrator
Guide for Avaya Communication Manager (Document # 03-300509). These features
can be administered, configured, and customized using the tools provided under the
Avaya Integrated Management application suite.
Offer Structure
The Avaya Integrated Management offers are designed to meet the varied needs of
a broad range of customers, from small businesses with one medium sized office, to
large businesses with thousands of users in a single location, as well as
corporations with distributed Communication Manager networks with multiple
branch offices.
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The following offers are available for Avaya Integrated Management Release 3.1:
¾
Standard Integrated Management
¾
Administration Tools
¾
Enterprise Network Management (and/or Network Management for Solaris)
¾
System Management
¾
Monitoring Management
Each offer is described in more detail below.
¾ The Standard Management Entitlement Offer ships with most Avaya
Communication Manager new systems and upgrades. It incorporates SNMP
agents, devices managers, a 90-day trial of Avaya Voice Over IP (VoIP)
Monitoring Manager, and one license for Avaya Site Administration (ASA)
which provide the core capabilities needed to administer a single voice
system.
¾ The Enterprise Management Entitlement Offer includes the items above plus
Voice Announcement Manager (VAM), and the Avaya Network Management
Console (NMC) with VoIP System View.
In addition to the standard
Management applications are
include value-added advanced
an enterprise to flexibly match
entitlements, additional optional Avaya Integrated
packaged within four specific offers; these offers
application options aligned by functionality, allowing
management capabilities to specific requirements:
¾ Administration Tools are an ideal solution for enterprises with a small
network of voice systems. The offer expands upon standard management by
adding a license for Avaya Site Administration. It also provides a solution for
managing adds, changes, backups, and broadcasts of recorded
announcements to voice systems over the enterprise network with Avaya
Voice Announcement Manager. For flexibility, additional Basic Administration
license options are available to meet the needs of larger campus
environments and multi-country deployments.
¾ Enterprise Network Management (and/or Network Management for Solaris) is
a super-set of Basic Administration Tools, encompassing the core applications
in all key areas of management. The offer is ideally suited for enterprises
with mid to large-scale branch office VoIP deployments. Enterprise Network
Management starts with the Network Management Console. The console is
the launch point for centralized management tools designed to simplify
provisioning/installation, access security administration, software and
firmware upgrades, and trouble-shooting tasks required to support a
distributed network of IP telephony locations. The offer also incorporates
SNMP agents, devices managers, and a single license each for Avaya Site
Administration and Voice Announcement Manager. Additional license options
provide flexibility to match a variety of network scenarios.
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¾ System Management can be added to any of the other Integrated
Management offers. It is designed for enterprises with multiple voices
systems from Avaya DEFINITY® Communications Server Release 9.5 up to
and including Avaya Communication Manager software. The offer includes
comprehensive traffic analysis and system information tools to trouble-shoot
and perform long term trending and performance base-lining of larger
networks, along with advanced Helpdesk functionality that provides
managers with key information for trouble resolution. This solution also
offers advanced management support for Enterprise Survivable Servers
(ESS), making it ideal for enterprises with high-availability requirements. It
is also designed for enterprises with distributed management groups that
desire advanced control over how they structure administrative roles and
access.
¾ Monitoring Management helps any enterprise deploying VoIP to better
manage the performance of the application through their converged network.
VoIP Monitoring Manager is designed as a standalone solution for use with
any Avaya Communications System Release 11 or higher.
Avaya Integrated Management is a software only offer. It is expected that VoiceCon
will supply the required hardware and operating system.
6.0.1 System/Port Capacity
Vendor Response Requirement
Identify the maximum number of independent IPTS communications systems that can be
supported by the proposed systems management server, and the maximum number of
user ports that can be passively and actively supported.
Avaya Response:
Included with this design for VoiceCon is the Standard Management Entitlement,
Standard Management Entitlement
The Standard Management entitlement for a new Communication Manager system
includes one license for Avaya Site Administration and a 90-day trial of VoIP
Monitoring Manager.
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Administration Tools Offer
¾
Basic Administration Tools
o
¾
Campus Administration Tools (Optional)
o
¾
Includes one license for Avaya Site Administration, one license for
Voice Announcement Manager, and a 90-day trial of VoIP Monitoring
Manager.
Includes unlimited licenses of Avaya Site Administration within a single
campus, unlimited licenses of Voice Announcement Manager within a
single campus, and a 90-day trial of VoIP Monitoring Manager
Enterprise Administration Tools (Optional)
o
Includes unlimited licenses for Avaya Site Administration in a single
country, unlimited licenses for Voice Announcement Manager in a
single country, and a 90-day trial of VoIP Monitoring Manager.
Enterprise Network Management Offer
¾
Enterprise Management Entitlement Upgrade
o
¾
Enterprise Management Offer (1-5 Managers Systems)
o
¾
Shipped with qualifying Communication Manager upgrades. Provides
for management of one Communication Manager voice system of up to
1,000 managed devices and 4,000 IP phones. Includes one license for
Avaya Site Administration, one license for Voice Announcement
Manager, one Server License for provisioning and network
management applications, and a 90-day trial of VoIP Monitoring
Manager.
Provides for management of a Communication Manager network by up
to 5 managers, with up to 1000 managed devices, and 4000 IP
phones. Includes five licenses for Avaya Site Administration, five
licenses for Voice Announcement Manager, one server license for
network management and provisioning applications, and a 90-day trial
of VoIP Monitoring Manager
Enterprise Management Solaris Offer (1-5 Managers Systems)
o
Provides for management of a Communication Manager network by up
to 5 managers, with up to 1000 managed devices and 4000 IP phones.
Includes five licenses for Avaya Site Administration, five licenses for
Voice Announcement Manager, one server license for Network
Management on Solaris, and a 90-day trial of VoIP Monitoring
Manager.
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Communication Manager System Management Offer
¾
System Management (1-10 voice systems)
o
¾
System Management (11-50 voice systems)
o
¾
Provides advanced monitoring and management capabilities to
enterprises with 1-10 voice systems. Includes Fault and Performance
Manager, Multi-Site Administration, Proxy Agent, and network
management system integration.
Provides advanced monitoring and management capabilities to
enterprises with 11-50 voice systems. Includes Fault and Performance
Manager, Multi-Site Administration, Proxy Agent, and network
management system integration.
System Management (51+ voice systems)
o
Provides advanced monitoring and management capabilities to
enterprises with 51+ voice systems. Includes Fault and Performance
Manager, Multi-Site Administration, Proxy Agent, and network
management system integration.
Monitoring Management Offer
¾
VoIP Monitoring
o
¾
Additional licenses for VoIP Monitoring Manager
o
¾
VoIP Monitoring Manager with License for 2000 endpoints and 40
gateways. Note: When counting gateways you must count any VoIP
engines and not physical gateways, so MedPro Boards and any other
Gateway with a VoIP Engine count as 1
Licensing for an additional 2000 endpoints and\or 40 gateways.
VoIP Monitoring - Enterprise Offer
o
Unrestricted license for VoIP Monitoring Manager within a single
enterprise location (no licensing limit on the number of IP Phones).
Note that customers may need multiple VoIP Monitoring Manager
servers to monitor their entire organization.
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6.0.2 Terminal Capacity
Vendor Response Requirement
Identify the maximum number of configurable and active PC client terminals that can be
configured as part of the proposed management server system.
Avaya Response:
The proposed solution includes five user licenses for the Avaya Integrated
Management applications.
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.
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.
Avaya Response:
The S8720 has an embedded SNMP native agent, which can send traps to the
Network Management Console. In addition, Integrated Management supports
integration with HP OpenView on Windows 2000 and Solaris. Integration with other
vendors Network Management Systems will be investigated for future releases of
the Avaya Integrated Management based on customer feedback. Using the SNMP
Native Agent VoiceCon can elect to send traps to 15 locations one of which could be
a NMS, they can also download the traps from support.avaya.com.
Avaya Network Management Console provides VoiceCon with a product that
can operate standalone as the your main Network Management System or can
integrate with the HP OpenView Network Node Manager 6.2/6.4 on the Windows
2000 platform and HP OpenView Network Node Manager 6.2/ 6/4 on Solaris 8/9.
The Network Management Console provides discovery of IP devices on the
customer’s network, fault monitoring of those voice systems, as well as many
applications that enable management of your network of devices. These
applications include the Avaya Network Management Console, Avaya Network
Configuration, Avaya Software Update Manager, Avaya QoS Manager, Avaya
Address Manager, Avaya VLAN Manager and optionally the Avaya SMON Manager.
The Avaya Network Management Console supports converged network
environments composed of multi-vendor equipment (This encompasses only major
devices from key vendors). The Avaya Network Management Console and
applications supports all Avaya IP based voice and data devices, wired and wireless,
to create a full convergence solution.
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The Network Management System Integration (NMSI) operates with the NMS
(Avaya Network Manager and/or a customer-provided HP OpenView) to provide
icons on the System view for various Avaya Communication Manager voice
systems, as well as providing fault status of the voice systems.
Feature Highlights – Web-based client support for Windows XP
Professional
¾
Integration with HP OpenView on Windows 2000 and on Solaris
¾
Security enhancements
¾
¾
¾
o
Support for SNMPv3
o
Login mode for secure access to applications
Wireless LAN management
o
Avaya Wireless AP Manager, supporting Avaya Wireless AP-3 (version
2.0 and later – up to version 2.3), AP-4 and AP-5
o
AP-MON, providing performance and traffic analysis
New Hardware Support
o
Avaya P460
o
P882 10Gb and 48-port modules,
Avaya QoS Manager
o
¾
Defines and deploys policy rules for simplified QoS and access control
configuration
SystemView – logical view of VoIP device hierarchy
o
Reflects Avaya Communication Manager servers, LSPs, CLANs, Media
Gateways, and IP Phones
The proposed Avaya Softconsole supports LDAP compliant directories which are
managed through VoiceCon’s LDAP application directly.
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6.0.4 Security Features
Unauthorized access to the communications system is a major concern. The ability to
detect security problems is desirable beyond mechanisms to prevent security problems.
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?
Avaya Response:
All access to the S8720 Communication system requires a valid login on the
communication manager for each user to be able to access.
All products offered through the Integrated Management Suite follow all United
States guidelines for encryption algorithms that can be exported.
Linux Servers:
Although remote access is currently required through dial-up access over a modem,
security precautions are available on the Linux server to ensure access by Avaya
Services is restricted to only information that is required. This can be achieved by
use of standard Linux operating system capabilities of creating Avaya Services only
login/password sets and permission groups allowing only read and/or write access
to only the partitions of the server that are needed by Avaya Services.
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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Avaya Response:
Comply.
The proposed Avaya Media Servers and Gateways include embedded native agents.
The Modular Messaging application is managed with a proxy agent, web-based
device managers, as well as Avaya Site Administration and Avaya Voice
Announcement Manager. The offer includes MultiSite Administration, Fault and
Performance Manager, Voice Announcement Manager, VoIP Monitoring Manager,
and Network Management Console. The capabilities listed above are supported via
these applications, as described below. The following optional applications enable
you to administer site moves, perform adds and changes, configure distributed
networks and campus environments, and manage performance in converged
networks.
Avaya Enterprise Network Management Web Pages
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Avaya Site Administration
Avaya Site Administration (ASA) supports individual Avaya Media Servers and
Avaya messaging platforms. ASA is a PC-based Windows (2000/XP Professional)
tool that presents a graphical user interface along with special task wizards that
simplify basic administration functions by reducing repetitive tasks for
moves/adds/changes, finding extensions, monitoring, and a variety of other
functions. A graphical tool provides access to more advanced administrative tasks.
Using ASA, an administrator can access the switch for administration and
maintenance purposes; this can be done remotely from any point on the network,
including offsite assuming VPN access is in place so the administrator can log into
the switch. A secure connection to the voice and messaging systems can be made
through SSH. ASA allows the administrator to setup, configure, and troubleshoot
any station on the network from the centralized administration platform.
Below are some features of ASA:
¾
Administration Wizards including Add/Change/Remove User and Change User
Extension
¾
Terminal emulation and GEDI (Graphically Enhanced DEFINITY Interface) for
access to Communication Manager for other administrative operations
¾
Templates for adding new objects
¾
Native Name Entry for Station Object
¾
Import/Export of administration fields for select objects
¾
Report generation
¾
Schedule tasks to run at a later time
¾
Button Label Printing
¾
Basic Fault & Performance capabilities
¾
Email Notification (for Monitoring and Report Generation)
¾
Call Center: Option by Agent, Location Preference Distribution, and Vector
Enhancements
¾
OPTIM/SCCAN
¾
Large Scale Meet-Me Conferencing
¾
System Capacity Wizard
¾
Support for ASG
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The ASA interface contains some very valuable applications
administrators to customize features and monitor system parameters:
that
allow
¾
General Tab
¾
Administrators can manage users, stations, ports, and extensions.
¾
Advanced Tab
¾
Administrators can create templates, import and export data, and start
emulations.
¾
Fault & Performance Tab
¾
Administrators can establish data polling schedules, and monitor trunks and
alarms.
ASA General Tab
Avaya Communication Manager offers over 700 standard features, each offering
enterprise-class functionality and usability. ASA provides comprehensive tools to
manage users, stations, ports, and extensions; through ASA, the System
Administrator is able to enable/disable features at the user level. ASA also provides
a web based tool for basic feature changes that may need to be done when access
to the full ASA tool is not practical.
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ASA General Tab – Tree View
With ASA, the following General functions are available and customizable:
¾
This helps users create a new extension on voice system, or a new mailbox
on the messaging system.
¾
Start GEDI: This opens a GEDI (Graphically Enhanced DEFINITY Interface)
window. Click this icon to administer the voice system when there is no
wizard to help.
¾
This lets users change information associated with a particular extension on
the voice system (or mailbox on the messaging system).
¾
This helps users delete a person from the voice system and messaging
system. It automates the process of removing that person from all the
groups they may be a part of.
¾
This automates the process of adding a bridged appearance to a telephone.
Users might want to add a bridged appearance so someone can pick up calls
that are going to a different phone, with the press of a button.
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¾
This enables users to see the dial ranges that have been set up in the dial
plan for the selected voice system. Users might want to use this wizard
before adding an object to the voice system, so they don’t accidentally assign
the new object an extension that has been saved for a different use.
¾
This displays a list of the stations that have been defined on the voice
system, based on the selection criteria (extension range and/or set type)
that users specify.
¾
This finds the next unused port. Users need this information before they can
add certain objects (like new phones) to the voice system.
¾
This finds the next unused extension, once users give it the extension range
they want to look in. Users might want to use this wizard before adding an
object to the voice system, so they don’t accidentally assign the new object
an extension that has already been used.
¾
This enables users to print labels for the buttons on your telephones, using a
button label template that was set up beforehand.
¾
This lets users "swap stations," or move a station to the other station’s port
and vice versa.
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ASA Advanced Tab
ASA Advanced Tab - Tree View
With ASA, the following Advanced functions are available and customizable:
¾
This enables users to create template copies from the most common objects
that are added to the voice system (like phones, hunt groups, and so on).
Templates save users from entering the same information in many places,
and can help increase the consistency of settings from one object to another.
¾
This enables users to use the templates created with the "Create New
Template" wizard to add new objects to the voice system.
¾
This enables users to export data from the voice system to a file that their
call accounting software can use.
¾
This enables users to generate and schedule a DEFINTY® command report
that can be printed, exported, or e-mailed.
¾
This enables users to export data from the voice system to a file that other
software applications (like spreadsheets or databases) can use.
¾
This enables users to import data from a spreadsheet or database application
into the voice system.
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¾
This enables users to search the voice system for certain settings or
information and optionally replace the old information with new information.
¾
This enables users to start a terminal emulation session to the voice system.
Terminal emulation gives users access to screens they can use to administer
the system.
ASA Fault and Performance
Through the Fault and Performance Tab, ASA provides basic fault and performance
capabilities. Alarm notification can be made to the system administration platform
as well as to an operator console (or any station) by simply programming a feature
access code to indicate minor or major alarms. ASA also provides email notification
for monitoring and report generation.
ASA Fault and Performance Tab - Tree View
With ASA, the following Fault and Performance functions are available and
customizable:
¾
Data Polling Schedule: This displays a screen that shows a schedule and
status of all data polling tasks for a particular voice system. From that
screen, administrators can set up polling tasks.
¾
Monitor Alarms: This enables administrators to set up ASA to automatically
notify them of any unresolved alarms on the system.
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¾
Monitor Trunks: This enables administrators to set ASA up to automatically
check for trunks that are out of service and notify them when it finds one.
¾
Trunk Analyzer: This enables administrators to gather trunk data, from which
they can create a report that gives recommendations on achieving a specified
grade of service.
¾
Processor Occupancy: This enables administrators to measure and assess the
performance of their voice system over a period of time under different
loads. After the data is collected, administrators can then plot that data on a
graph.
¾
Call Traffic: This enables administrators to monitor the volume, type, and
distribution of call traffic in order to measure how a voice system is
performing. After the data is collected, they can then plot that data on a
graph.
¾
System Capacity: This enables administrators to view the maximum
capacities of their system and their current level of usage.
¾
Audits: This enables administrators to look for redundant or unused data on
their voice system. It can help them clean up their voice system quickly
without having to manually look through a large number of screens.
¾
Time Synchronization: This enables administrators to set up the voice system
time to match the clocks on network PCs, or set an offset between the voice
system time and PC time in the event they are in different time zones.
¾
Hardware Manager: This enables administrators to see a representation of
their voice system configuration in a tabular and carrier layout. Included in
this representation are all of the voice system’s boards and carriers, and any
alarms or errors for that hardware.
Avaya MultiSite Administration (MSA) - Detailed Description
Avaya MultiSite Administration (MSA) is a client-server based application that
provides multi-user, graphical, web-based management of more than one Avaya
media server and media gateway. It provides centralized management of
distributed networks and campus environments.
MSA Features
Avaya MultiSite Administration offers these powerful features:
¾
Enables multiple administrators to administer the same (or separate) Avaya
media servers at the same time, remotely.
¾
Offers graphical station and system administration screens.
¾
Offers easy-to-use wizards for basic administration tasks.
¾
Allows users to (using terminal emulation) administer other telephony
devices.
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MSA Toolbar
The toolbar displays icons for commonly performed tasks. The first six icons are
determined by MultiSite Administration. The icons to the right of the first six can be
customized. The toolbar can contain a maximum of 20 icons.
¾
Toolbar Customizer: Clicking this icon allows users to customize the toolbar.
¾
System Properties: Clicking this icon allows users to view the "system
properties" associated with the given voice system, including information like
name, IP address, and so on.
¾
System CutThrough: Clicking this icon launches the CutThrough window,
which is a terminal emulation window that enables users to log into the voice
system.
¾
View Logs: Clicking this icon allows users to see a history of general activity
on the MultiSite Administration server, and activity on the voice system that
has been initiated by a MultiSite Administration user.
¾
Station Wizard: Clicking this icon starts the Station Wizard, which allows
users to perform various station administration tasks.
¾
View Scheduled Entries: Clicking this icon lets users view the reports and
commands that are currently scheduled to occur on the given voice system.
Fault and Performance Manager (FPM)
Avaya Fault and Performance Manager (FPM) provides a network map or system
view of the converged network and the tabular tools to monitor the status and
performance of the devices on the enterprise’s network. The user can see into the
network to examine faults and performance data from the Avaya Media Servers on
the network.
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FPM collects configuration, fault, and performance data from Avaya Proxy Agent or
directly from an IP-enabled voice system using OSSI, and then displays the data in
text, tables, and graphic formats.
The device data and user accounts are stored in the Avaya Integrated Management
Database (IMD) and shared by the Avaya Fault and Performance Manager and the
Avaya MultiSite Administration and Proxy Agent applications.
FPM operates on versions of Red Hat Enterprise Linux. It is available in the Avaya
Integrated Management Release 3.0, System Management offer.
FPM Features
The primary features of the Fault and Performance Manager are:
¾
Graphical User Interface (GUI): The main window provides the following
views of the managed nodes in the network, such as, system groups, DCS
trunk connectivity, IP trunk connectivity, and clusters.
¾
Configuration: The users can view the configuration and administered
properties of all supported systems (managed nodes) in both a graphic view
and a table view.
¾
Administration: Users can define the system-wide parameters for the
features, such as, data collection, exception logging, exception filtering, and
exception alerting.
¾
Report Manager: Users can define the parameters for individual reports for
all or selected systems. The report options include performance,
configuration, and exceptions. The users can view the reports on screen in
both the table and chart formats or direct the reports to a printer, HTML file,
GIF file, or ASCII file.
¾
Scheduled Reports: The users can schedule reports to run on a daily, weekly,
or monthly basis, and edit and delete schedules as needed.
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Integrated Management Database
Integrated Management Database provides a single interface for administration of
users and configuration data of managed systems for MultiSite Administration, Fault
and Performance Manager, and Proxy Agent. Integrated Management Database is
an integrated management application with HTML pages viewable in Internet
Explorer 6.0 on Windows 2000, XP and Windows 2003
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New Password Security Features:
¾
Force Admin Login Password Change (First use of default password for the
“admin” login shall automatically prompt the user to change the password)
¾
Login password requirements (Passwords must be at least 8 characters in
length; Length is administrable)
¾
Password Re-Use (Passwords cannot match any one of the previous four
passwords)
¾
Password Contents (Passwords must contain at least one alphabetic
character and one numeric character)
¾
Password Change Rate Limiter (Passwords cannot be changed more than
once a day; Rate is administrable)
¾
Password Change Error Display (display error messages that provide
guidance on why a password was not accepted)
¾
Password-Aging (passwords must be changed after 60 days; Cycle length is
administrable)
New Password Security Configuration and Display
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Part 1 – Section 6 - Systems Management
Avaya Proxy Agent
Avaya Proxy Agent is a protocol conversion resource that operates on a Red Hat
Linux platform. Proxy Agent uses TCP/IP ports to collect configuration and
management data from supported systems. It converts OSSI data generated by
Avaya Call Processing software and Avaya Communication Manager into SNMP data.
Avaya Proxy Agent and Avaya Fault and Performance Manager work together to
provide a view of the health and performance of the network systems.
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The Proxy Agent generates SNMP traps when supported systems generate alarms
and system errors. Proxy Agent then transmits the SNMP data to Fault and
Performance Manager on the System Management Server. The System
Management Server is a Linux system. Avaya Proxy Agent is available in the Avaya
Integrated Management Release 3.0, System Management offer.
Avaya Voice Announcement Manager (VAM)
Avaya Voice Announcement Manager (VAM) is a stand-alone application. It is
installed on a Microsoft Windows client PC and provides the ability to add, change,
and remove recorded announcements and transfer them to Avaya media servers.
Using WAV files, an announcement can be stored and sent over a LAN without file
conversion. The WAV file repository simplifies storage and management of WAV
files.
Users can view the current status of announcements at any time. Announcement
files can be copied, backed up, or restored from Avaya media servers and gateways
via the LAN. They can also schedule broadcasts and backups of announcements to
single or multiple VAL boards and/or Media Servers.
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Avaya VoIP Monitoring Manager
The Enterprise Network Management offer also includes a complementary 90-day
trial of Avaya VoIP Monitoring Manager. VoIP Monitoring Manager is a QoS
monitoring and feedback software tool that supports real-time visualization of VoIP
network performance from a client GUI application residing on the LAN or via
remote access. VoIP Monitoring Manager receives Real-Time Protocol Control
Protocol (RTCP) packets from IP phones and VoIP engines on media gateways. This
allows the network manager to monitor a variety of call paths in real-time,
including:
¾
Endpoint-to-Endpoint (end-to-end through the network)
¾
Endpoint-to-Gateway
¾
Gateway-to-Gateway (IP trunk monitoring over the WAN).
VoIP Monitoring Manager provides information on all QoS parameters using an
easy-to-understand graphical meter display. To support quick identification and
localization of network issues that impact voice quality, VoIP Monitoring Manager
can also generate traps based on preconfigured thresholds to any Network
Management System.
The standards-based architecture of Avaya Integrated Management tools comply
with accepted standards such as Simple Network Management Protocol (SNMP).
This open architecture not only encompasses a wide range of Avaya communication
devices and server, but also extends to integration with HP OpenView and products
from Extreme Networks via integration with the Extreme Networks EPICenter
network management application.
The VoIP Monitoring application is a diagnostic tool that monitors the quality of
voice transmissions over the network. This product tracks transmissions of Real
Time Control Protocol (RTCP) packets. The product listens to RTCP packets and
populates the database with historical data.
You can query the database from a web browser to generate reports and to view
real-time graphics of the RTCP packet transmission. The reports show active
phones and other IP endpoints and identify the IP endpoints that are experiencing
problems with jitter, delay, and packet loss.
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The Avaya VoIP Monitoring Manager shown below provides the ability to monitor
VoIP network quality, by receiving QoS statistics from Avaya IP end-points and
providing the means to visually display the real-time data.
VoIP Monitoring Manager
Beyond trouble-shooting and monitoring, VoIP Monitoring Manager is also a proactive notification solution that warns the network manager of potential degradation
of voice quality in their network – providing valuable time to identify, localize, and
fix issues in the network as they arise. In this scenario, VoIP Monitoring Manager
can be configured to automatically send SNMP (Simple Network Management
Protocol) traps to a Network Management System (NMS) based on a number of
QoS threshold policies.
SMON Manager provides traffic monitoring designed for networks that incorporate
the point-to-point transmission schemes of Avaya Ethernet and gigabit Ethernet
switches. It provides a top-down view of your network traffic that drills down to
individual switches, VLANs, and ports. Administrators can display voice endpoints,
view all enterprise alarms and performance history reports, customize alarms and
thresholds, and tie alarms and thresholds to alerts—all with no effect on switch
performance. For VoIP trouble-shooting, a VoIP port monitoring feature works with
Avaya VoIP Monitoring Manager’s real-time call session information to help network
managers more quickly identify, diagnose, and fix network IP telephony issues by
filtering network information to display only traffic on ports that have connected IP
phones.
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Avaya Fault and Performance Manager provides a hierarchical view of the health
and status of all voice systems in the network, with drill-down capabilities to display
information on configuration, fault, and health data of individual systems and
adjuncts in text, tabular, and graphical formats. For enterprises planning the
deployment of fault-tolerant networks, Fault and Performance Manager includes
support for monitoring of Enterprise Survivable Servers (ESS) and the management
of port network moves between servers.
Fault and Performance Manager is also a powerful reporting tool, with system-wide
administration of parameters that allow the network manager to quickly define
settings for data collection, logging, and alert levels, and a flexible report manager
that provides detailed information on performance, configuration, and
exceptions/alarms. To help simplify interpretation of system messages and alarms
as they appear, an integrated helpdesk feature automatically launches the
Communication Manager maintenance manual with the specific alarm in focus. The
ability to set up filtering of alarms on a system wide or individual system basis
makes finding alarms and elevating the alert status a valuable trouble shooting
tool. Alerts can be easily emailed or forwarded on to any SNMP compliant
application
Fault and Performance Manager
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Avaya Network Management Console
The Avaya Network Management Console shown above provides discovery of IP
devices including voice elements such as Avaya S8300 Media Server, Avaya S8720
Media Server, TN2501AP boards and IP phones placing the discovered devices on a
network map, as illustrated below.
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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.
Avaya Response:
All above functions and more are supported. Avaya Integrated Management will
allow VoiceCon to administer business site moves, adds and changes; configure
distributed networks and campus environments; and manage performance in
converged networks. Administrators can use the LAN to transfer recorded
announcements to Avaya Media Servers and Gateways, and managing a network of
multiple spare processors over ATM wide area networks - it enhances system
reliability and business continuity.
The Avaya Station Administration Wizard is accessible from the Standard
Management Solutions home page on the media server.
With this easy-to-use wizard, you can quickly add, move, or change data on a
station.
Main Features
¾
Add station – Adds a new station to your system.
¾
Change station – modifies station parameters for a selected station.
¾
Move station – Gives the selected station a new extension and removes the
old extension.
¾
Find station – Queries the system for a station or range of stations.
¾
Find unused extensions – Queries the system for a list of available station
extensions.
¾
Each task includes additional steps to execute other features of the task.
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6.1.1 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.
Avaya Response:
Comply. All of the group definitions listed above are supported by the system
administration applications. Descriptions of the features are provided in Appendix 1,
Avaya Communication Manager Feature Descriptions.
6.1.2 Facilities Performance Management & Reports
The management system must be able to collect, analyze, and provide reports for a variety of
system operations.
Avaya Response:
Read and understood.
6.2.1 Basic Trunk Usage and Traffic
Trunk traffic records should be kept for all inbound and outbound calls, identifying the trunk group
and trunk channel, time and duration of call.
Vendor Response Requirement
Confirm that the proposed facilities management system satisfies this requirement.
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Avaya Response:
Comply. The standard reporting capabilities of Avaya Communication Manager
include extensive traffic reporting capabilities on trunk groups. The reports are
included in the base application, and do not require additional servers or software.
There are several reports available for trunk groups, including:
¾
The Trunk Group Summary report gives traffic measurements for all trunk
groups except for personal central office line groups. By using this report,
you can determine the trunk group total usage (in CCS), the total number of
calls, trunk blockage, and other measurement data.
¾
The Trunk Group Hourly report provides data necessary to validate the
information in the Trunk Group Summary report and to size the trunk
groups. A separate report is generated for each trunk group.
¾
The Trunk Out of Service report lists up to a maximum of five trunks (in each
trunk group) out of service when sampled. The number of times the trunks
are out of service when sampled is also given. The trunk outage data is kept
for the current day, the previous day, and the last hour.
¾
The Trunk Group Status report gives a current indication of the load on
various trunk groups in terms of the number of calls waiting to be serviced.
For each trunk group, the Trunk Group Status Report displays the number of
calls in the queue waiting to be serviced. For comparative analysis, the trunk
members in the group active on calls are also displayed. With this data, it is
possible to rearrange the members in the groups to provide load balancing.
For example, if one group shows a higher number of calls waiting in the
queue and the size of the group is too small; more members can be added to
that group.
¾
The Call-by-call (CBC) Trunk Group Measurements report displays last-hour
traffic data for any specified Call-by Call trunk group, provided the trunk
group had a Usage Allocation Plan (UAP) administered for the last-hour. Use
the report to monitor the trunk group and to determine if the UAP meets
current needs. Whenever it is determined changes are required, you must
make these changes on the appropriate trunk group screen(s).
¾
The Trunk Lightly Used report lists the five trunk members with the lowest
number of calls carried for each trunk group. The trunk lightly used data is
kept for the current day, the previous day, and the last hour.
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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).
Avaya Response:
Comply. The Trunk Group Summary report gives traffic measurements for all trunk
groups except for personal central office line groups. By using this report, you can
determine the trunk group total usage (in CCS), the total number of calls, trunk
blockage, and other measurement data.
Trunk Group Summary report field descriptions
Field
Description
Peak Hour for
All
Trunk
Groups
The hour during the specified day with the largest total usage, when
summed over all trunk groups. Peak hour and busy hour are
synonymous. With conventional traffic theory data analysis, there are
two methods for determining the peak hour. One is the time-coincident
peak hour, meaning that hourly usage values are averaged across days
for each hour of the day. The other is the bouncing peak hour, meaning
that the highest usage is selected for each day without regard to the
average across days. For the bouncing peak hour the highest load on a
given day may or may not occur during the time-coincident busy hour.
These traffic reports and accompanying trunk group data worksheet only
use the bouncing peak hour method. Note that if the total usage for the
current hour equals the total usage for the previous peak hour, then the
peak hour is the hour with the greatest number of total seizures.
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Field
1
Description
Grp No.
Group Number. A number that identifies each trunk group associated
with the displayed data. Group numbers are displayed in numerical
order, beginning with the lowest administered number and continuing to
the highest administered number.
Grp Siz
Group Size. The number of administered trunks in the trunk group.
Grp Type
Group Type. The type of trunk in the trunk group. The system
monitors/measures the following trunk types:
Access Tie Trunk (Access)
Advanced Private Line Termination (aplt)
Central Office (co)
Public Network Service Customer Provided Equipment (cpe)
Direct Inward Dialing (did)
Direct Inward/Outward Dialing (diod)
Digital Multiplexed Interface Bit Oriented Signaling (dmi-bos)
Foreign Exchange (fx)
Integrated Services Digital Network (isdn-pri)
Release Link Trunk (rlt)
Tandem (tan)
Tie Trunk (tie)
Wide Area Telecommunications Service (wats)
Grp Dir
Trunk Group Direction. Identifies whether the trunk group is incoming
(inc), outgoing (out), or two-way (two).
Meas Hour
Measurement Hour. The hour (using 24-hour clock) in which the
measurements are taken. For the last-hour report, it is the last hour of
measurement (each trunk group’s measurement hour is identical; but
not necessarily the same as the indicated peak hour for the day). For
the today-peakreport, the measurement hour is the peak hour for each
trunk group thus far today (each trunk group’s measurement hour could
be different). For the yesterday-peakreport, the measurement hour is
the peak hour for each trunk group yesterday (each trunk group’s
measurement hour can be different).
Total Usage1
Total usage (in CCS) for all trunks in the trunk group. Represents the
total time the trunks are busy (with calls) during the one-hour
measurement period. Total usage measures each time a trunk is seized
for use by an incoming call (whether it is picked up or not) or an out
going call (only after digits have been dialed).
Total Seize
The number of incoming and outgoing seizures carried on the trunk
group. This includes the number of times a trunk in the group is seized,
including false starts, don’t answer, and busy.
Inc. Seize
Incoming Seize. The number of incoming seizures carried on the trunk
group.
Grp Ovf
Group Overflow. The number of calls offered to a trunk group not
carried or queued (if a queue is present). Calls rejected for authorization
reasons are not included.
The usage that wideband calls contribute to this measurement is proportional to the resources the calls consume. For example, a 384-kbps call
contributes six times more to the total usage than does a 64-kbps call.
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Field
Description
Que Siz
Trunk Group Queue Size. A number (0 to 100) that identifies the
number of slots assigned to the trunk group queue. This number
represents how many calls may be held in queue by the trunk group. If
0 is displayed, then no queue is administered. Hence, the other queue
measurements are also 0. Generally, the queue size should be larger
than the trunk group size; however, not more than three times as large
as the trunk group size.
Call Qued
Calls Queued. The total number of calls that entered the trunk group
queue after finding all trunks busy.
Que Ovf
Queue Overflow. The total number of calls not queued because the
queue is full. These calls receive a reorder signal. Suggested actions:
Generally, this field indicates the number 0. If this field indicates a high
number, then either the queue size may be too small, or add more
trunks to reduce the number of calls queuing.
Queue Abandoned. The number of calls removed from the queue in one
of the following manners:
By the system because they have been in the queue for more than 30
minutes
Que Abd
By the user (for example, dialing the cancel code).
Suggested action: Typically, this field indicates a small number.
However, a large number generally indicates the queue size is too large
and people are abandoning because they remained in queue for a long
holding time and gave up.
Out Srv
December 1, 2006
Out of Service. The number of trunks in the trunk group out of service
(listed as maintenance busy) at the time data is collected. An individual
trunk may be taken out of service by the switch whenever an excessive
number of errors occur or by maintenance personnel to run diagnostic
tests. Suggested action: If the trunks are removed from service by the
switch, then the appropriate maintenance personnel should be notified.
The objective is to keep all members of a trunk group “in service.”
Generally, you should not make adjustments to the trunk group because
of “Out of Service” trunks, but should get those trunks returned to
service.
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Field
Description
Percentage All Trunks Busy. The percentage of time all trunks in the
trunk group were simultaneously in use during the measurement
interval. In use means the trunks are busy— either serving calls NOTE:
or because they are busied-out by maintenance.
Suggested actions:
% ATB
1. If the group direction is outgoing or two-way, then a high number in
the % ATB field and nothing in the Grp Ovfl or Que Ovfl indicates
everything is functioning normally. However, a more typical scenario
is a high number in this field and a high number in the Grp Ovfl field.
This indicates a possible problem that necessitates further analysis.
Unless it is the last trunk group in the pattern, overflow is to the
next choice trunk group, and the number in the Grp Ovfl field is of
no great significance. Otherwise, the obvious choice is to add more
trunks to the trunk group.
2. If the group direction is incoming, then a high number in this field is
bad. It indicates some incoming calls are probably blocked.
Generally, you want to add more trunks, thus lowering the % ATB
and decreasing the number of calls blocked.
% Out Blk
Percentage Outgoing Blocking. The percentage of offered calls not
carried on the trunk group. It does not include unauthorized calls
denied service on the trunk group (due to restrictions) or calls carried
on the trunk group but do not successfully complete at the far end
(that is, where there is no answer). For trunk groups without a queue,
the calls not carried are those calls that arrive when all trunks are
busy. The number of Outgoing Seizures is calculated as follows:
Similarly, the equation for calculating Outgoing Calls Offered is as
follows:
For trunk groups with a queue, the calls not carried are those calls that
arrive when all trunks are busy and the queue is full (Queue Overflow)
and calls removed from queue before being carried (Queue
Abandoned). For this scenario, the Percentage Outgoing Blocking is
calculated as follows:
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Field
Description
Suggested actions:
You can increase the length of the queue rather than adding more
trunks. Subsequently, you should monitor the Que Abd field to insure it
stays within reasonable limits.
If conditions are such that Step 1 is not appropriate, then you may find
it necessary to add more trunks.
Wideband Flag
If the trunk group supports wideband (n X DS0) switching, a “W” appears next to
the trunk group entry. In addition, if any trunk group on the report supports
wideband switching, the tag“W = Wideband Support” appears in the report
heading.
6.2.1.2 Outgoing Trunk Route Counters
Vendor Response Requirement
Confirm that outgoing trunk route counters measure and report: Number of outgoing attempts;
Number of successful calls overflowing to another route; Number of lost calls due to blocking;
Number of blocked trunks in measurement; Traffic intensity (Erlangs).
Avaya Response:
Comply. Several outgoing trunk route measurements are available across various
reports; please refer to the complete description provided in item 6.2.1.1.
6.2.1.3 Incoming Trunk Route Counters
Vendor Response Requirement
Confirm that incoming trunk route counters measure and report: Number of incoming call
attempts; Number of trunks in the measurement; Number of blocked trunks in the measurement;
Traffic intensity (Erlangs).
Avaya Response:
Comply. Several outgoing trunk route measurements are available across various
reports; please refer to the complete description provided in item 6.2.1.1.
6.2.1.4 Both Way Trunk Route Counters
Vendor Response Requirement
Confirm that both way trunk route counters measure and report: Number of incoming call
attempts; Number of trunks in the measurement; Number of blocked trunks in the measurement;
Traffic intensity (Erlangs).
Avaya Response:
Comply. Please refer to the report and description provided in item 6.2.1.1.
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6.2.2 Attendant Consoles
Attendant counters should measure all attendants in the system, or individual attendant positions.
Record measurements include: number of answered calls; number of calls initiated by attendant;
accumulated handling time for all calls; accumulated handling time for recalls; accumulated
handling time for calls initiated by attendant; accumulated total delay time for recalls; number of
answered recalls; number of abandoned attendant recalls; accumulated waiting time for
abandoned calls to an attendant; accumulated waiting time for abandoned recalls, and
accumulated response time for all types of calls.
Vendor Response Requirement
Confirm that attendant counters measure and provide reports for each of the listed parameters.
Identify attendant parameters which are not measured.
Avaya Response:
Comply. The proposed solution includes inherent detailed attendant reporting. No
additional hardware or software is required to support the feature. The attendant
group reports are used to assess the quality of service provided customers calling
through the listed directory numbers and to facilitate the management of the
attendant group so it is neither under – nor over-staffed. Attendant group
measurements appear on two reports.
¾
The Attendant Group report provides hourly traffic measurements for the
attendant group as a whole.
¾
The Attendant Positions report gives peak individual attendant position
measurements.
Both reports are available as PEAK reports for yesterday’s peak hour, today’s peak
hour, and the last hour. A peak hour is the hour within a 24-hour period with the
greatest usage (Time Talk plus Time Held) for the specified day. Hourly data for the
entire attendant group can be obtained by polling the Attendant Group Report on
an hourly basis.
Attendant Group Measurements Report
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Attendant Group Measurements report field descriptions
Field
Description
Grp Siz
Group Size. The number of attendant positions (consoles)
administered for the groups.
Meas Hour
Measurement Hour. The hours represented are indicated by the
labels in the right-hand column (YEST PEAK — the hours of
yesterday’s peak activity, TODAY PEAK— today’s peak activity,
and LAST HOUR— the last hour activity).
Calls Ans
Calls Answered. The number of calls answered by all active
attendants during the measurement hour. With Total Usage and
Calls Answered, you can determine the Average Work Time
AWT), which is the time it takes an attendant to handle a call.
Calls placed to individual attendant extensions or that route to
an attendant via a hunt group do not increment the Calls Ans
counter.
Calls Aband
Calls Abandoned. The number of calls that ring an attendant
group and drop (the caller hangs up) before an attendant
answers. Where applicable, this total includes calls abandoned
from the attendant queue before answered. A call abandoned
after placed on hold is not included in this measurement,
because it is already added to the calls answered measurement.
Calls Qued
Calls Queued. The total number of calls placed in the attendant
queue (delayed) because no attendants are available. Calls
remain in the queue until one of the following occurs:
An attendant becomes available and the call is connected.
The caller, while waiting in the queue, abandons the call (hangs
up) before an attendant is available.
The call covers to another point in a coverage path.
Calls H-Abd
Calls Held-Abandoned. The number of calls that abandon while
the caller is in hold mode. Held calls which time out and re-alert
are included in the held-abandoned call count.
Calls Held
Calls Held. The number of calls answered by the attendant group
and subsequently placed on hold by the attendant group.
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Field
Description
Time Available. The time during which the “pos avail” lamp is lit
on all attendant consoles, and the attendants are not talking on
calls but are available to handle new calls. Measured in Centum
(hundred) Call Seconds (CCS).
NOTE:
An attendant can have calls on hold and still be
available. For example, if two attendants are available for 15
minutes each during the measurement hour, the total available
time would be 30 minutes or 18 CCS (0.5 hour X 36 CCS per
hour).
Time Avail
Consoles may be administered either with their own unique
extension number or without any extension number. For the
“with extension number” case, traffic measurements for
outgoing calls and incoming calls to the extension are allotted to
the console’s extension number and not to the attendant group.
For the “without” case, all traffic measurements are allotted to
the attendant group. The time the console is on outgoing calls is
not included in the attendant group’s Time Avail measurement.
Attendants are not available and do not accumulate time
available when:
The position is in Night Service
The position was busied-out
The headset is unplugged
The attendant is servicing a call
Also referred to as talk time. The total time, during the
measurement interval, attendant(s) are active or talking on a
loop (measured in CCS).
Talk time is not started until the call is answered by the
attendant. The duration of time between the call terminating at
the attendant console and when the call is answered is not
accumulated as either Avail Time or Talk Time.
Time Talk
Calls split by the attendant do not accumulate talk time from the
point when the attendant presses the start button until the call is
placed.
Calls routed to an attendant via a hunt group are treated as calls
to the attendant extension and therefore do not accumulate talk
time. An attendant can have up to six calls on hold at any one
time. However, each attendant can only be active on one loop at
a time.
Time Held
December 1, 2006
Also referred to as held time. The total amount of time
(measured in seconds) the attendants have calls on hold.
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Field
Description
Also referred to as time to abandon. The average amount of time
calls spend in queue and/or ringing at the console before the
callers hang up (measured in seconds).
Note: If the Time Abnd value is smaller than the Speed Ans
(sec) value, you need more agents. As a contrast, if the Time
Abnd value is larger than the Speed Ans (sec) value, the
attendant group should process the calls faster. The attendant
group should be engineered so Time Abnd approximately equals
the calculated average delay.
Time Abnd
Speed of Answer. The average elapsed time from when a call
terminates at the attendant group to when the call is answered
by an attendant (measured in seconds). The average time calls
wait to ring an attendant (Queue Usage / Calls Answered). The
Queue Usage is the total time calls spend in the attendant
queue.
Speed
(Sec)
Ans Note: Calls terminate either directly to an attendant console and
subsequently begin ringing or in the attendant queue when there
are no attendant positions available.
If the average time to abandon is equal to or exceeds 9999
seconds, the value 9999 displays in the field.
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Attendant Positions Measurements report
The Attendant Positions Measurements report provides hourly individual attendant
position measurements. It is used to assess personnel performance, and to identify
when additional training may be necessary.
Attendant Positions Measurements Report – page 1
Attendant Positions Measurements Report – page 2
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Attendant Positions Measurement report field descriptions
Field
Description
Attd ID
Attendant ID. A number between 1 and the maximum number of
attendants to identify which attendant’s data is being displayed.
This number is chosen by the user upon administering this
attendant.
Time Talk
The time the attendant is active on calls (in CCS); measured from
the time the attendant activates an attendant loop until the loop
is released. If more than one loop is active on an attendant
console at one time, the usage is counted only once (for
example, one attendant is not counted as being busy more than
once at a single time).
Time Held
The time the attendant had calls on hold (measured in seconds).
Time Avail
Time Available. The total time the subject attendant is available
to receive calls during the polling interval (measured in CCS).
Calls Ans
Calls Answered. The total number of calls answered by this
attendant (measured in CCS). Calls placed to an individual
attendant extension or that route to an attendant via a hunt
group do not increment the Calls Ansfield.
Attendant Speed of Answer report
The Attendant Speed of Answer report gives the console attendant group average
speed of answer for each hour of a 24-hour period, for either yesterday or today.
Attendant Speed of Answer report — page 1
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Attendant Speed of Answer report — page 2
Attendant Speed of Answer report field descriptions
Field
Description
Meas Hour
Measurement Hour. The starting time (using the 24-hour clock)
of the hour during which the data was recorded.
Average Speed
of Answer(sec)
A graphic display of the average time taken by attendants to
answer calls.
Speed Ans
(sec)
Speed of Answer (in seconds). The average speed of answer is
also displayed numerically in seconds for each hour in the report
interval.
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.
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Avaya Response:
Comply. Avaya VoIP Monitoring Manager, in combination with the reports resident
on the server, provides all of the functions listed above.
6.2.4 Traffic distribution
When applicable, traffic distribution across the internal switching network should be measured for
each local TDM bus, traffic over each highway bus, and traffic across the center stage switch by
each switch network interface link.
Vendor Response Requirement
Confirm that traffic distribution is measured and reported for each switch network element listed.
Identify what is not measured and reported.
Avaya Response:
Comply. Traffic distribution is measured by the VoIP Monitoring application
described elsewhere in this proposal.
6.2.5 Busy hour traffic analysis
Busy hour traffic analysis measurements for trunks, stations, and the internal switch network
should be performed and reported for any one hour interval for any time of the day.
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.
Avaya Response:
Comply. Please refer to the busy hour reporting capabilities discussed in detail
elsewhere in this proposal.
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.
Avaya Response:
Comply. Erlang ratings are provided on the trunk group reports discussed in
elsewhere this proposal.
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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
Avaya Response:
Comply. The proposed Avaya solution supports several processor occupancy
reports, defined as the percentage of time the configuration’s processor is busy
performing call processing tasks, maintenance tasks, administration tasks, and
operating system tasks. As a contrast, the percentage of time the processor is not
used is referred to as idle occupancy.
The primary objectives of the processor occupancy reports are:
¾
To provide a summary of customer usage data so processor occupancy and
available capacity can be determined.
¾
To display, on a per time interval basis, the processor occupancy and
associated calling rates which facilitates the isolation of certain customer
reported problems.
There are four different processor occupancy commands:
¾
list measurements occupancy summary
¾
list measurements occupancy last-hour
¾
list measurements occupancy busiest-intervals
¾
list measurements communications-links
The first three commands provide processor occupancy data and associated call
traffic for different measurement intervals. The last command provides a picture of
the traffic data generated on each processor interface link.
6.2.8 Threshold Alarms
For a variety of system hardware devices it should be possible to define a congestion threshold
value, and measure generated alarms. Alarms are recorded in an Alarm Record Log. The types
of devices that can be tracked include: tone receivers; DTMF senders and receivers; conference
bridges; trunk routes; modem groups.
Vendor Response Requirement
Confirm recording and reporting of alarms for each listed item.
Avaya Response:
Comply. Alarms are logged in the Hardware Error Report and can be reviewed on a
regular basis. The administrator can view all active system errors on the error log.
The user can also specify a particular component of the system or a certain time
period to be reported on the error log.
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This example shows errors on trunk routes and digital line cards; however, all error
types are tracked.
6.2.9 Feature Usage
Feature usage counters for selected station features, e.g., call forward, call transfer, add-on
conference, and attendant system features, e.g., recall, break-in, should be measured and
reported for programmed intervals.
Vendor Response Requirement
Confirm recording and reporting of feature usage counters for both station and attendant
operations.
Avaya Response:
Comply. There are several reports that provide measurement information about
features. Including the following:
The following reports provide information about the Announcements feature:
¾
Denial Events Log – if a working VAL announcement file is deleted over FTP,
the next attempt to play the announcement fails, and the system adds a
software event to the Denial Events Log. Users can view the Denial Events
log to see if the announcement was deleted, and to see if other events
occurred that are related to announcements.
¾
Announcement Measurements – users can view a report of announcement
measurements. This report includes how many times an announcement was
queued to play, how many callers dropped while in the queue, and how many
times all announcement ports were busy during the report period.
The ACA Measurements Report (ACA) shows the audit trail for ACA calls.
The following reports provide information about the Call Coverage feature:
¾
The Coverage Path Measurement report shows coverage activity about the
coverage paths.
o
The Principal Coverage Measurement report shows coverage activity
about the called extensions.
o
The Call Detail Recording (CDR) report shows the outgoing trunk calls.
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The following reports provide information about the Enhanced 911 feature:
¾
The Emergency Access Calls report shows the following information for each
emergency call:
o
Extension
o
Event
o
Type of call
o
Time
The following reports provide information about the Security Violation Notification
feature:
¾
The Security Violations Status Report shows details of the last sixteen
violations of each type. The Barrier Code and Authorization Code reports also
include the calling party number from which the attempt was made, if that
information is available.
The following reports provide information about the Station Hunting feature:
¾
The List Usage report shows all the extensions that use an extension as the
hunt-to-station.
The Uniform Dial Plan report shows the details of the Dial Plan.
6.2.10 VoIP Monitoring
The management system should collect and store data to track usage and performance data of
IP gateway devices, IP phones, and VoIP intercom/trunk calls. VoIP information reports may
include: tracking of IP gateway devices and calls that pass through each gateway; gateway
congestion; assignment of services or routes to gateways; tracking of phone numbers dialed or
originating off-site numbers; and IP gateway addresses.
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.
Avaya Response:
Avaya VoIP Monitoring provides the requested features, as described elsewhere in
this proposal. In addition, the proposed solution includes six reports that list the
activity on your IP media processor circuit packs, useful for managing both intraswitch traffic, as well as IP trunking. These reports are described below:
¾
IP Codec Resource Hourly report - lists the codec resources used on all IP
media processors for the last 24 hours, from the current hour backwards, for
a specific region. This report lists separate information for the G.711 codecs
and the G.723/G.729 codecs.
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IP Codec Resource Hourly report
Field
Range
Description
Meas Hour
0000-2300
Measurement Hour. The hour that the data was collected, from the
current hour backward.
Region
1-44
The network region of the IP media processors being measured.
The region number is assigned on the Ip-interfaces screen during
switch administration.
DSP Rscs
0-9999
Digital Signaling Processor Resources. Total number of IP codec
resources, or voice channels, in the region.
G711
(Erl)
0-9999
Amount of time (in erlangs) that G.711 codecs were in use during
the measurement period. The time is measured from the time the
voice channel is allocated until it is released; including the time
that the voice channel is on a call.
Usage
This measurement is calculated by adding the total time (in
seconds) that G.711 resources on all IP media processors are in
use, divided by 3600.
G711 In Reg
Peg
0-65535
The total number of times an IP media processor port in the region
was allocated to a G.711 call.
G711 Out of
Reg peg
0-65535
The total number of times an IP media processor port was needed
in the region for a G.711 call, but was successfully allocated to a
resource in another region.
If the “Region” fields on the Inter Network Region Connection
Management screen are blank, then this measurement will always
be 0.
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Field
Range
G723/9
Usage (ERL)
0-9999
Description
Amount of time (in erlangs) that G.723 or G.729 codecs were in
use during the measurement period. The time is measured from
the time the voice channel is allocated until it is released; including
the time that the voice channel is on a call.
This measurement is calculated by adding the total time (in
seconds) that G.723 or G.729 resources on all IP media
processors are in use, divided by 3600.
G723/9
Reg peg
In
0-65535
The total number of times an IP media processor port in the region
was allocated to a G.723 or G.729 call.
G723/9 Out
of Reg peg
0-65535
The total number of times an IP media processor port was needed
in the region for a G.723 or G.729 call, but was successfully
allocated to a resource in another region. If the “Region” fields on
the Inter Network Region Connection Management screen are
blank, then this measurement will always be 0.
IP Codec Resource Summary report - lists the codec resources used on all IP media
processors for a specific peak hour for all regions. You can list reports for
yesterday’s peak, today’s peak, or the last hour. This report lists separate
information for the G.711 codecs and the G.723/G.729 codecs.
IP Codec Resource Summary Report
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Field
Range
Description
Meas Hour
0000-2300
Measurement Hour. The hour that the data was collected.
Region
1-44
The network region of the IP media processors being measured.
The region number is assigned on the IP Interfaces screen during
switch administration.
DSP Rscs
0-9999
Digital Signaling Processor Resources. Total number of IP codec
resources, or voice channels, in the region.
G711 Usage
(Erl)
0-9999
G711 In Reg
Peg
0-65535
G711 Out of
Reg peg
Amount of time (in erlangs) that G.711 codecs were in use during
the measurement period. The time is measured from the time the
voice channel is allocated until it is released; including the time that
the voice channel is on a call.
This measurement is calculated by adding the total time (in seconds)
that G.711 resources on all IP media processors are in use, divided
by 3600.
0-65535
The total number of times an IP media processor port in the region
was allocated to a G.711 call.
The total number of times an IP media processor port was needed in
the region for a G.711 call, but was successfully allocated to a
resource in another region.
If the “Region” fields on the Inter Network Region Connection
Management screen are blank, then this measurement will always
be 0.
Amount of time (in erlangs) that G.723 or G.729 codecs were in use
during the measurement period. The time is measured from the time
the voice channel is allocated until it is released; including the time
that the voice channel is on a call.
G723/9
Usage (ERL)
0-9999
G723/9 In
Reg peg
0-65535
The total number of times an IP media processor port in the region
was allocated to a G.723 or G.729 call.
0-65535
The total number of times an IP media processor port was needed in
the region for a G.723 or G.729 call, but was successfully allocated
to a resource in another region. If the “Region” fields on the Inter
Network Region Connection Management screen are blank, then
this measurement will always be 0.
G723/9 Out
of Reg peg
This measurement is calculated by adding the total time (in seconds)
that G.723 or G.729 resources on all IP media processors are in
use, divided by 3600.
IP Codec Resource Detail report - lists the codec resources used on all IP media
processors for a specific peak hour for a specific region. You can list reports for
yesterday’s peak, today’s peak, or the last hour. This report lists separate
information for the G.711 codecs and the G.723/G.729 codecs.
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IP Codec Resource Detail report
Field
Range
Description
Meas Hour
0000-2300
Measurement Hour. The hour that the data was collected.
Region
1-44
The network region of the IP media processors being
measured. The region number is assigned on the Ipinterfaces screen during switch administration.
DSP Rscs
0-9999
Digital Signaling Processor Resources. Total number of IP
codec resources, or voice channels, in the region.
G711 Usage
(Erl)
0-9999
Amount of time (in erlangs) that G.711 codecs were in
use during the measurement period. The time is
measured from the time the voice channel is allocated
until it is released; including the time that the voice
channel is on a call.
This measurement is calculated by adding the total time
(in seconds) that G.711 resources on all IP media
processors are in use, divided by 3600.
G711 In Reg
Peg
G711 Out of
Reg Peg
G723/9
Usage (Erl)
0-65535
0-65535
0-9999
The total number of times an IP media processor port in
the region was allocated to a G.711 call.
The total number of times an IP media processor port was
needed in the region for a G.711 call, but was successfully
allocated to a resource in another region.
If the “Region” fields on the Inter Network Region
Connection Management screen are blank, then this
measurement will always be 0.
Amount of time (in erlangs) that G.723 or G.729 codecs
were in use during the measurement period. The time is
measured from the time the voice channel is allocated
until it is released; including the time that the voice
channel is on a call.
This measurement is calculated by adding the total time
(in seconds) that G.723 or G.729 resources on all IP
media processors are in use, divided by 3600.
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Field
Range
Description
G723/9 In
Reg Peg
0-65535
The total number of times an IP media processor port in
the region was allocated to a G.723 or G.729 call.
G723/9 Out
of Reg Peg
0-65535
The total number of times an IP media processor port was
needed in the region for a G.723 or G.729 call, but was
successfully allocated to a resource in another region.
If the “Region” fields on the Inter Network Region
Connection Management screen are blank, then this
measurement will always be 0.
IP DSP Resource Hourly report - lists the codec resources used on all IP media
processors for the last 24 hours, from the current hour backwards, for a specific
region.
IP DSP Resource Hourly report
Field
Range
Description
Meas Hour
0000-2300
Measurement Hour. The hour that the data was collected.
Region
1-44
The network region of the IP media processors being
measured. The region number is assigned on the Ipinterfaces screen during switch administration.
DSP Rscs
0-9999
Digital Signaling Processor Resources. Total number of IP
codec resources, or voice channels, in the region.
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Field
DSP
(Erl)
Usage
In Reg Peg
Range
0-9999
Description
Digital Signaling Processor Usage in Erlangs. Amount of
time (in erlangs) that all codecs were in use during the
measurement period. The time is measured from the time
the voice channel is allocated until it is released; including
the time that the voice channel is on a call.
This measurement is calculated by adding the total time (in
seconds) that G.711 resources on all IP media processors are in
use plus twice the total time (in seconds) that G.723 and G.729
resources are in use plus twice the time (in seconds) that fax relay
resources are in use, divided by 3600.
0-65535
The total number of times an IP media processor port in the region
was allocated to a call
Out of Reg
Peg
0-65535
The total number of times an IP media processor port was
needed in the region for a call, but was successfully
allocated to a resource in another region. If the “Region”
fields on the Inter Network Region Connection
Management screen are blank, then this measurement
will always be 0.
Denied Peg
0-65535
The total number of times an IP media processor port was
needed in the region for a call, but all media ports in all
regions were busy and the call did not go through.
0-99
Percentage Blocked. The percent of attempted use of IP
media processor ports in the region that were not
successful (blocked). This percent includes calls that were
denied after they were successfully allocated out of the
region.
% Blk
% out of Srv
0-99
Percentage Out of Service. The percent of CCS time that
any IP media processor ports were out of service during
the measurement period. This percent includes ports that
were manually busied out or maintenance busy during the
measured interval.
This measurement is calculated by multiplying by 100 the
total time (in CCS) that any port was out of service
divided by the number of available resources times 36
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IP DSP Resource Summary report - lists the codec resources used on all IP media
processors for a specific peak hour for all regions. You can list reports for
yesterday’s peak, today’s peak, or the last hour.
IP DSP Resource Summary Report
Field
Range
Description
Meas Hour
00002300
Measurement Hour. The hour that the data was collected.
Region
1-44
The network region of the IP media processors being measured. The
region number is assigned on the Ip-interfaces screen during switch
administration.
DSP Rscs
0-9999
Digital Signaling Processor Resources. Total number of IP codec
resources, or voice channels, in the region.
Digital Signaling Processor Usage in Erlangs. Amount of time (in
erlangs) that all codecs were in use during the measurement period. The
time is measured from the time the voice channel is allocated until it is
released; including the time that the voice channel is on a call.
DSP Usage
(Erl)
0-9999
In Reg Peg
0-65535
Out of Reg
Peg
0-65535
Denied Peg
0-65535
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This measurement is calculated by adding the total time (in seconds)
that G.711 resources on all IP media processors are in use plus twice
the total time (in seconds) that G.723 and G.729 resources are in use,
plus twice the time (in seconds) that fax relay resources are in use,
divided by 3600.
The total number of times an IP media processor port in the region was
allocated to a call
The total number of times an IP media processor port was needed in the
region for a call, but was successfully allocated to a resource in another
region.
If the “Region” fields on the Inter Network Region Connection
Management screen are blank, then this measurement will always be 0.
The total number of times an IP media processor port was needed in the
region for a call, but all media ports in all regions were busy and the call
did not go through.
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Field
% Blk
% Out of
Srv
Range
0-99
0-99
Description
Percent Blocked. The percent of attempted use of IP media processor
ports in the region that were not successful (blocked). This percent
includes calls that were denied after they were successfully allocated out
of the region.
Percent Out of Service. The percent of CCS time that any IP media
processor ports were out of service during the measurement period. This
percent includes ports that were manually busied out or maintenance
busy during the measured interval.
This measurement is calculated by multiplying by 100 the following:
Total time (in CCS) that any port was out of service divided by the
number of available resources times 36.
IP DSP Resource Detail report – lists the codec resources used on all IP media
processors for a specific peak hour for a specific region. You can list reports for
yesterday’s peak, today’s peak, or the last hour.
Field
Range
Description
Meas Hour
0000-2300
Measurement Hour. The hour that the data was collected.
Region
1-44
The network region of the IP media processors being measured. The
region number is assigned on the Ip-interfaces screen during switch
administration.
DSP Rscs
0-9999
Digital Signaling Processor Resources. Total number of IP codec
resources, or voice channels, in the region.
Digital Signaling Processor Usage in Erlangs. Amount of time (in
erlangs) that all codecs were in use during the measurement period. The
time is measured from the time the voice channel is allocated until it is
released; including the time that the voice channel is on a call.
DSP Usage
(ERL)
0-9999
In Reg Peg
0-65535
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This measurement is calculated by adding the total time (in seconds)
that G.711 resources on all IP media processors are in use plus twice
the total time (in seconds) that G.723 and G.729 resources are in use,
plus twice the time (in seconds) that fax relay resources are in use
divided by 3600.
The total number of times an IP media processor port in the region was
allocated to a call
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Field
Range
Description
Out of Reg
peg
0-65535
The total number of times an IP media processor port was needed in the
region for a call, but was successfully allocated to a resource in another
region. If the “Region” fields on the Inter Network Region Connection
Management screen are blank, then this measurement will always be 0.
Denied Peg
0-65535
The total number of times an IP media processor port was needed in the
region for a call, but all media ports in all regions were busy and the call
did not go through.
0-99
Percent Blocked. The percent of attempted use of IP media processor
ports in the region that were not successful (blocked). This percent
includes calls that were denied after they were successfully allocated out
of the region.
% Blk
% Out of
Srv
0-99
Percent Out of Service. The percent of CCS time that any IP media
processor ports were out of service during the measurement period. This
percent includes ports that were manually busied out or maintenance
busy during the measured interval.
This measurement is calculated by multiplying by 100 the following:
Total time (in CCS) that any port was out of service divided by the
number of available resources times 36.
6.3
Optional Reports
Directory records may include each subscriber’s name along with a variety of phone numbers
such as primary, published, listed, emergency, and alternate, as well as authorization code
information, job title, employee number, current employment status and SSN.
Inventory records and management is used to administer any kind of inventory product part,
including: PBX common equipment (cabinets, carriers, circuit cards); voice terminals and module
options; jacks, and button maps. The reports allow administrators to accurately re-charge items.
Inventory can be tracked by data such as user, system (PBX or other networks), jack, serial
number, asset tags, trouble calls, recurring and non-recurring costs, and general ledger
codes. The inventory management system may also include records containing the following
data: purchase date, purchase order number, depreciation, lease dates, manufacturer and
warranty information.
Cabling records keep track of all cable, wire pairs, distribution frames, wiring closets and all
connections (including circuits) down to both the position and the pair level. Cable records include
starting and ending locations, description, type and function. Individual cable lengths are
maintained and automatically added, as is the decibel loss, for the entire path. Information can
also be provided on the status of all cable runs, as well as the number of pairs it contains, the
status of the pairs, and the type of service it provides.
Vendor Response Requirement
Identify and briefly describe your proposed management system’s Directory, Inventory, and
Cabling reports, if available.
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Avaya Response:
The Avaya Software Update Manager is optional, but not requested elsewhere. It
helps network managers ensure that all devices are running with the current
version of the device software.
The product downloads the latest update software and you just point and client to
complete the update. You also receive automatic notices when new releases are
available.
You can also view a complete inventory of all Avaya devices residing on the
network, including: IP address, device name, and software version.
Main Features
¾
Automatically connects with the Avaya web site and downloads new releases.
¾
Highlights devices not running the latest software version.
¾
Simultaneously downloads images to multiple devices.
¾
Downloads on-demand or schedules downloads for off-peak hours.
¾
Supports G700 and G350 devices.
¾
Supports all new families and new devices.
The optional Avaya Address Manager contains network maintenance tools that
network managers use to locate IP addresses or hosts within the network.
The tool displays a centralized list that contains hosts discovered in the network
and correlates between IP address, MAC address, and device port connectivity.
Main Features
¾
Locates switch ports that are related to specific hosts.
¾
Discovers duplicate IP addresses or port policy violations.
¾
Generates, prints, and export reports.
Additional Optional Reporting Capabilities
Avaya Network Management Console
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Avaya VoIP Monitoring Manager
Avaya VoIP Monitoring Manager is a QoS monitoring and feedback Windows 2000
software tool that allows the customer to visualize the real-time operation of VoIP
systems. The tool provides information on QoS parameters related to VoIP quality.
Avaya VoIP Monitoring Manager provides the ability to view QoS related information
through a client GUI application from the customer’s LAN or via remote access.
Avaya VoIP Monitoring Manager is also capable of being configured to generate
traps, associated with VoIP QoS, sent to any NMS. Avaya VoIP Monitoring Manager
can receive RTP Control Protocol (RTCP) packets from the following entities: Avaya
IP telephones, Avaya IP soft phones, VoIP engines (on media gateways) and
prowler boards. The RTCP data for current calls is published in the Real-time
Transport Protocol (RTP) Message Information Base (MIB) via the SNMP agent
running on the server. The historical RTCP data is published in the VMON MIB. Both
forms of RTCP data can be viewable by either field support, a system administrator
or services personnel.
Avaya VoIP Monitoring Manager is integrated with the Avaya SMON Manager, such
that Avaya VoIP Monitoring Manager can also be launched from Avaya Network
Management Console, when the Avaya Network Management Console is standalone
or integrated with a customer-provided version of HP OpenView. The Avaya VoIP
Monitoring Manager will also operate standalone without the Avaya Network
Management Console or a customer-provided version of HP OpenView.
6.4.0 Call Detail Recording
Call Detail Record (CDR) data should be compiled for all successful incoming and outgoing trunk
calls. Call record fields typically include the following:
•
Date
•
Time
•
Call Duration
•
Condition Code (categorizes information represented in the call record)
•
Trunk Access Codes
•
Dialed Number
•
Calling Number
•
Account Code
•
Authorization Code
•
Facility Restriction Level for Private Network Calls
•
Transit Network Selection Code (ISDN access code to route calls to a specific inter-exchange
carrier)
•
ISDN Bearer Capability Class
•
Call Bandwidth
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•
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.
Avaya Response:
Avaya Site Administration can generate call accounting data that is needed by call
accounting and billing systems. The following categories can be placed into an
export file that can be imported into a third party billing system: agent login-id,
authorization codes, station data, trunk-groups and circuits.
The Call Detail Recording feature of Communication Manager records detailed call
information on incoming and outgoing calls for the purpose of call accounting, and
sends this call information to a Call Detail Recording (CDR) output device. You can
specify the trunk groups and extensions for which you want records to be kept as
well as the type of information to be recorded. You can keep track of both internal
and external calls. This application contains a wide variety of administrable options
and capabilities.
Two types of formats are sent to the CDR output device, date record and call detail
formats. You can use the customized record format to make up your own call
record. You can determine the data elements you want and their positions in the
record. This method may be necessary if you want to include certain data elements
that are not available on the standard formats. However, whatever device you use
to interpret the CDR data needs to be programmed to accept these formats.
Consult your Avaya representative before using a custom record format.
The following data elements are available:
Data Field Description
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Time of day-hours
Authorization code
Dialed number
Time of day-minutes
Feature flag
Calling number
Duration-hours
Access code used
Account code
Duration-minutes
Outgoing circuit ID FRL
Duration-tenths of minutes Access code dialed
IXC
Condition code
Incoming circuit ID
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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
Confirm support of each listed maintenance monitoring activity. Identify any activity not
supported.
Avaya Response:
Comply. All maintenance and monitoring activities listed above are supported. It is
important to understand that internal systems diagnostics play an important role in
the reliability of a system. Avaya Communication Manager has more than 30
percent of its lines of code dedicated to internal diagnostics to maintain constant
vigilance over the health and well being of the system. It is these diagnostics that
fundamentally give the Avaya solution its advantage over many other systems
because in conjunction with these diagnostics comes a patented system of artificial
intelligence we call Avaya EXPERT SystemsSM Diagnostic Tools.
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.
Avaya Response:
The Communication Manager application monitors and logs alarms. Alarms are
classified as major, minor, or warning, depending on the degree of severity and the
effect on the system.
The system keeps a record of every alarm that it detects. This record, the alarm
log, and the error log can be displayed locally on the management terminal. An
alarm is classified as MAJOR, MINOR, or WARNING, depending on its effect on system
operation. Alarms are also classified as ON-BOARD or OFF-BOARD.
¾
MAJOR alarms identify failures that cause critical degradation of service and
require immediate attention. Major alarms can occur on standby components
without affecting service, since their active counterparts continue to function.
¾
MINOR alarms identify failures that cause some service degradation but do
not render a crucial portion of the system inoperable. The condition requires
attention, but typically a minor alarm affects only a few trunks or stations or
a single feature.
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¾
WARNING alarms identify failures that cause no significant degradation of
service or failures of equipment external to the system. These are not
reported to the Avaya alarm receiving system or the attendant console.
Alarms are further classified as:
¾
on-board problems originate within the circuitry of the alarmed circuit pack
¾
off-board problems originate in a process or component that is external to
the circuit pack
Feature button lamps on any phone can be administered to act as alarm indicators,
similar to the alarm lamp on the attendant console.
6.5.2 Maintenance Reports
Vendor Response Requirement
Identify any and all available maintenance alarm reports provided by your management system.
Avaya Response:
If a maintenance object in the system begins to fail some of the periodic tests, the
system automatically generates an alarm that indicates the system needs to be
restored to a normal condition.
Alarms are communicated to the system users and technicians by entries in the
alarm log and the lighting of LEDs located on the attendant console, on all circuit
packs, on the server, on the Cajun Ethernet switch (if used), and, optionally, on
telephones designated by VoiceCon’s System Administrator. Warning alarms are not
reported to the attendant console.
The system alarm log is available for viewing via the administrative terminal. In
addition, Avaya Site Administration provides an alarm monitoring capability, as
illustrated below:
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6.5.3 Remote Maintenance
Vendor Response Requirement
Briefly describe the available options used to support remote maintenance operations for both
customer access and for an outside maintenance service provider. Specify how the system alerts
a remote service center when an alarm condition occurs, the trunk circuit requirements for alert
transmissions, and security measures to prevent unauthorized access.
Avaya Response:
When an Avaya Communication Manager enabled system recognizes not just
outright failures, but any system anomaly that might be service affecting, the
application performs a “cry for help” by out calling to the Avaya Remote Global
Technical Services (GTS) center. When the call is received a trouble ticket is
created while the alarm in question is automatically routed to EXPERT Systems
Diagnostic Tools to begin trouble resolution. With millions of hours of actual inservice information from the largest, longest installed base of telephony platforms
in North America, EXPERT Systems patented algorithms are able to make decisions
on how to resolve the alarm condition and in combination with the GTS Tier III
engineers, EXPERT Systems clears over 98% of all system-generated alarms
remotely.
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This service is provided during warranty for any equipment covered under an Avaya
maintenance contract purchased at point of sale. To our knowledge, this capability
has not been replicated by any other vendor in the market and most other suppliers
provide remote reactive monitoring, which in most cases is a substantially
additional cost reducing their overall “value” to VoiceCon.
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.
Avaya Response:
A fully integrated single source-provisioning tool is currently not available.
However, Avaya’s various administration tools offer many time and cost saving
benefits. These tools include the Avaya site Administration (ASA), Multi-System
Administration (MSA), the Avaya Provisioning and Installation Manager (PIM), and
third party applications using Avaya’s Directory enable Management application.
Each provides specific capabilities that will benefit any enterprise.
The ASA and MSA tools, as previously described, provided template and wizard
capabilities to allow for easy addition of users. This includes the addition of voice
mailboxes and would incorporate pre-designed images to help expedite the
installation process.
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Avaya Provisioning and Installation Manager (PIM)
PIM can help an enterprises save time and money on large-scale branch office
deployments by replacing error-prone and time-consuming device-by-device
provisioning with a tool that enables the definition of customized templates that can
be defined centrally and applied to large groups of gateways at one time. Designed
to integrate as part of the staging and installation process, Provisioning and
Installation manager allows network experts at the staging center to quickly gather
information using Electronic Pre-Installation Worksheets that can then be imported
into the application to create configuration templates and individual device profiles
to record unique parameters for each gateway. These templates and profiles can
be stored in advance as part of the initial staging process, then applied to groups of
gateways as part of the bulk provisioning process during actual installation. The
result is large-scale branch office deployments performed faster, with less error,
and with reduced need for technical expertise at the installation site. Provisioning
and Installation Manager can also be used for ongoing administrative tasks that
require configuration changes to groups of gateways.
Pre-Staging and Bulk Provisioning of Branch Office Media Gateways
Avaya Provisioning and Installation Manager (PIM) supports:
¾
Template-based bulk provisioning
¾
Electronic Pre-Installation Worksheets with direct import into the application
to create configuration templates and device profiles
¾
Scheduled provisioning of gateways at pre-stage or during actual installation
¾
Time-consuming device provisioning is replaced with a bulk provisioning
process that is faster and less expensive
¾
Electronic Pre-Installation Worksheets can be directly imported into PIM,
enabling deployments to be performed faster and with error
¾
Advanced provisioning reduces the need for technical expertise at the
installation sites.
Third Party Applications
There are a variety of Avaya DevConnect partners who have developed and can
customize more enhanced “all-in-one” tools using Avaya’s Directory Enabled
Managements application that provide LDAP level support.
Avaya continue to evolve the Integrated Management Tool Suite to incorporate
greater and greater functionality and strives to reach an “all in one” solution in the
future. No specific roadmap is available for distribution.
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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
2000
3,000
Number of Ports
64
96
Hours of Storage
1000
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.
Avaya Response:
Comply. The Avaya Modular Messaging solution is a powerful IP- and standardsbased voice messaging platform designed for single- or multi-site global
enterprises, centralized or de-centralized. It offers exceptional scalability and an
extensive feature package of call answering and messaging capabilities. Messages
are accessible any time, anywhere from a wide array of access devices including
telephones, PC graphical user interfaces, and PDAs. Graphical user capabilities,
explained in detail in response to 7.4.4, provide a means of unifying voice, fax, text
and email messages to be viewable, created, and acted upon from within the MS
Exchange Outlook client. Optionally, Avaya Modular Messaging for Microsoft
Exchange is also available as a single mailbox solution whereby all voice, fax, text,
and email messages are stored within a single inbox.
Using the proposed Avaya Unified Communication Center Speech Access
application, mobile professionals can stay connected to customers, associates and
partners via speech commands, wireless devices, and web browser. They can get to
their most important information and communications functions through their
phones, PDAs, desktop PC, or laptop—choosing the device that is most suitable at
that time.
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We are proposing the Avaya Modular Messaging with an Avaya Message Storage
Server. This configuration contains Avaya Messaging Application Server (MAS) units
with a Windows operating environment and an Avaya Message Storage Server
(MSS) with a Linux operating environment. For increased security, performance,
control, and application management, the MSS is connected to the Avaya
Messaging Application Server through a private Ethernet local area network (LAN).
The MSS is designed so that it has no administratively activated offline mode. UCC
Speech Access consists of an industry-standard server with Avaya UCC Speech
Access software. All new installations of the Avaya Messaging Application Server
software reside on the Avaya-provided S3500 server platform running Microsoft
Windows 2003 Server Application Kit with Service Pack 1. All new installations of
the Avaya Modular Messaging Message Storage Server software reside on the
Avaya S3500 platform with Red Hat Enterprise Linux Version 4.0 as the operating
system.
The H.323 integration provides connectivity with the Avaya PBX over an IP network.
The connectivity between the Avaya Message Application Server (MAS) and the
Avaya PBX is achieved over an IP-connected trunk defined as ISDN-PRI-equivalent
tie lines. This integration passes call information and MWI using QSIG messages
tunneled in the H.323 packet.
Regarding capacities, the IP connection to the Avaya S8720 supports up to 200
ports and 16,200 users. Up to 15,000 GSM hours/3,000 G.711 hours is available on
the MSS. Up to 5 Avaya Modular Messaging Application Servers are supported in a
voice mail domain. A voice mail domain can serve a network of switches, provided
the Administrator ensures that the network uses a single switch as a gateway to the
voice mail domain. Using the voice mail domain construct, up to 500 networked
nodes can support up to 250,000 subscribers.
An Automated Attendant is a fully integrated Modular Messaging feature used for
routing callers who have reached the organization’s access number to their desired
extension number. The Automated Attendant interface greets callers and guides
them through the process of entering the extension number of the called
subscriber. Administrators can configure Automated Attendant to allow callers to
identify the called subscriber by spelling the subscriber’s name using touchtone
keys. The Automated Attendant interface also allows callers to reach designated
system operators provided that an active schedule exists for the covering
extension.
Caller Applications are a collection of menus and prompts that allow administrators
to extend those parts of the Modular Messaging TUI that are accessible to callers.
Using Caller Applications, administrators can extend the system Automated
Attendant and the Common Caller Interface functionality, depending on the
requirements of the organization. A Caller Application is not a mailbox and does not
require that a mailbox be created on the message store or be dedicated to its use.
A Caller Application contains one or more nodes, each of which can interact with the
caller and pass control to another node. Up to 120 Caller Applications are supported
per voice mail domain.
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Caller Applications are separate applications—such as complex Automated
Attendants, listen-only mailboxes, and bulletin boards—which can be designed
using a Microsoft Windows graphical user interface (GUI)-based editor tool that is
deployed across voice mail domains. Caller Applications can be used to accomplish
most of the same functions as Automated Attendants (including nested Automated
Attendants).
Some basic functions that a Caller Application can provide include:
¾ Transferring callers to a specified mailbox
¾ Allowing callers to record messages
¾ Sending messages to either a mailbox number or an e-mail address that is
configured as part of the Caller Application
¾ Providing directory assistance
functionality of the TUI
for
callers
to
use
the
dial-by-name
7.1.0 Support for Open Standards
Vendor Response Requirement
Describe voice messaging system’s support for open standards. List the clients that can be used
with your proposed solution. For proprietary clients, detail minimum hardware and software
requirements
Avaya Response:
Avaya Modular Messaging supports the following industry open standards:
¾
Intel processors
¾
Dialogic Tip/Ring boards, Dialogic T1 and E1 port boards, and Dialogic Digital
Set Emulation (DSE) port boards. The proposed IP integration does not
require port cards; however, these are possible integration methods.
¾
Linux operating system (in Avaya MSS configurations) and Microsoft Windows
operating system (Avaya MAS)
IP and Internet Standards – These include IP for server-to-server transport, IP
Networking, IMAP4 and POP3 client access to messages (available in other
configurations but not required for the proposed solution), SMTP/MIME for sending
and receiving messages, and LDAP for attribute storage (user and system data) and
directory queries (name, number, address).
Switch Integrations – These include H.323-based IP integration, Q.Signaling
(QSIG), Enhanced Inband Analog, RS232 for serial switch integrations
(SMSI/SMDI), and Digital Set Emulation (DSE).
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Part 1 – Section 7 - Integrated Messaging System
Because Avaya has developed Modular Messaging as a standards-based platform,
significant flexibility is introduced with regards to switch integrations developed and
supported. This supported mix contains IP-based integrations, digital technology
(including T1/E1 and digital set emulation), and a variety of analog-based
technologies. Additionally, there are numerous integrations written, but not yet
generally available based on customer demand. Future development will also be
based on customer need and demand, as well as technological advances in this
area of development. At any given time, any combination of these integrations can
be supported within a customer's Modular Messaging network, combining many of
the developed integrations to support a broad mix of switching environments
throughout the country and world.
Audio Encoding Formats – These include Global System for Mobile Communications
(GSM) and G.711 (A-law and µ-law).
Other Standards – These include standards established by the government and
standards bodies for compliance areas. These include Product Safety, Electro
Magnetic Compliance (EMC), and Telecommunications.
Supported Clients
Avaya Modular Messaging includes IMAP4 access to messages from user client
software packages. Avaya has conducted successful interoperability testing with
Microsoft Outlook 2002, Microsoft Outlook 2000, Microsoft Outlook Express 5.0, IBM
Lotus Notes R6, and IBM Lotus Notes R5.
7.1.1 Security Features
Vendor Response Requirement
Describe security features available with the voice messaging system to prevent abuse and
unauthorized access.
Avaya Response:
Comply. Avaya is acutely aware of the need for stringent security measures to be in
place to minimize vulnerability. These areas include user mailboxes, system
administration procedures, and limiting caller transfer capabilities.
Mailbox Security
The Administrator initially configures user’s mailboxes with a default password or a
different randomly generated password for each new mailbox. On the first login
from the telephone user interface, the user can be required to set a new numeric
password between the Administrator-defined minimum and 15-digit maximum
length. The System Administrator establishes the minimum password length as a
system wide parameter. The password cannot match the mailbox number, include
leading zeros, match up to the previous 20 passwords, contain consecutive digits,
or use all the same digits. As an extra security measure, the Administrator can
choose to force users to press [#] after entering their mailbox passwords. Access to
messages via the Microsoft Exchange Graphical User Interface – Subscriber
Options, is controlled by the security scheme established for the client. Passwords
changed from the GUI are displayed as asterisks (*) as an added security measure.
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Users are encouraged to change their passwords often and can do so at any time
from either the TUI or via the graphical user interfaces. Passwords can be set by
the Administrator to expire every 1 to 999 days from the day they were last
changed. A certain number of previous passwords (specified by the Administrator)
may not be used. The user must change an expired password before being allowed
to send or retrieve messages through the telephone user interface. No one can
access a user's password, as passwords are not displayed on any administration
form. To allow for forgotten passwords, the Administrator has the ability to set a
new default password that the user must reset upon login from either of the
interfaces.
As an added security measure, the system locks the user’s mailbox after reaching
the system parameter limit of consecutive unsuccessful login attempts,
disconnecting the caller. This is tracked across multiple calls. The System
Administrator must then unlock the mailbox for the user.
System Administration Security
A Modular Messaging solution offers multiple administration security features to
protect the system. The Mailbox Manager Log In window allows the Administrator to
control access to Mailbox Manager with password protection. Upon initially starting
Mailbox Manager, the log in window includes two choices: Support Technician and
System Administrator. Mailbox Manager also allows the Administrator to create
additional login names and passwords for which the Administrator can restrict
access to certain types of records, known as Operators. Operators can modify their
own passwords, after they log in to Mailbox Manager under their own subscriber
names.
Voice Mail Domain Security
Access control lists can be edited to limit who has access to administration
applications and tools in a voice mail domain. These lists define the following types
of administration:
¾ System Administration – Members of the System Administration access
control list can access and use all Modular Messaging Software administration
applications and tools except subscriber administration tools. The default
System Administration access control list has a single entry containing the
account under which the first Messaging Application Server was installed.
¾ Subscriber Administration – Members of the Subscriber Administration access
control list can use subscriber administration tools to enable subscribers to
use Modular Messaging Software. The default list is empty.
Message Server Security
To enable the Messaging Application Server to log onto the Message Storage
Server, one or more account names and credentials for the Message Storage Server
must be configured.
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Limiting Caller Transfer Capabilities
One of the biggest sources of toll fraud against businesses is through hackers
transferring out of a voice mail system to an outside—usually long distance—
number. The system protects against unauthorized transfers through call transfer
controls that help prevent a caller from transferring to an outside line or number.
Toll restrictions can be implemented in the telephone system that restricts access
to the lines used by the Modular Messaging server for call processing.
The Administrator can select the digits that callers are allowed to enter as the initial
digit of an invalid mailbox number. This helps prevent toll fraud by prohibiting
callers from obtaining an external line when dialing the initial digital of an invalid
mailbox number.
The system can disconnect or transfer callers who make too many navigation
errors, including entering invalid mailbox numbers (systemwide parameter between
0 and 9).
7.2.0 Voice Mail Features
7.2.1 Forwarding
The system must provide access for forwarded calls from:
•
Customer telephone system
•
Direct central office (Business or Centrex lines)
•
800 Service lines
Vendor Response Requirement
Confirm support for each forwarding requirement.
Avaya Response:
Comply. Modular Messaging will accept calls forwarded from all three forwarded call
scenarios.
7.2.2. Disconnect Detection
The system should detect that a caller has hung up and immediately disconnect and restore the
line to service.
Vendor Response Requirement
Confirm support for this operation.
Avaya Response:
Comply.
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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.
Avaya Response:
Comply. Callers who know the extension number can enter it or access the Dial-ByName feature. If less than ten names match the letters entered, those names are
spoken in the form of a single digit menu and the caller chooses the correct
recipient.
If greater than nine matches are found, the system will request the caller to enter
more letters.
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:
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.
Avaya Response:
Partially comply. The proposed solution complies with all items specified above,
except for programming a separate greeting for calls received internally or
externally on the system. With the Call Handling feature, users can administer the
system to differentiate between calls that reach a mailbox because an extension is
busy or because there is no answer. Due to extensive research into providing the
distinction between internal and external calls on the Modular Messaging telephone
user interface, Avaya found that the technology has moved past this early voice
mail feature. As more and more businesses have virtual and mobile workers,
“external callers” can be employees as well. We found that providing the ability to
distinguish between whether a line is busy (the greeting can indicate the user is in
the office that way) or unanswered (the greeting can indicate the user is out for the
door) provides information that is of greater use to all callers, whether fellow
employees or business clients.
For the remaining requirements, we fully comply.
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Personalized Greetings
Users can record and activate multiple personal mailbox greetings, which can be rerecorded and activated at any time. The Administrator sets the maximum length of
greetings and prompts, which can be up to 30 seconds if desired. As recorded
greetings only occupy the number of seconds recorded, this automatically frees disk
space for other uses.
As an additional benefit, Modular Messaging uses industry standards to format and
store messages, VoiceCon can determine which storage method is used. If GSM is
selected, it provides “cell phone quality” encoded at an economical 13Kbps. As a
result, a voice message that is one minute long will require approximately 95.2 KB.
If a personal greeting is not recorded, the system plays a standard greeting with
the user’s recorded name and gives callers the option to leave a message.
Following are the types of greetings that can be made available to the user.
¾ Personal Greeting for All Calls
¾ Extended Absence Greeting – Callers cannot override listening to an Extended
Absence greeting. After listening, the caller is offered the options of entering
another extension, leaving a message, or returning to the Automated
Attendant menu. The Extended Absence greeting overrides all Call Handling
and Intercom Paging options. The user is reminded that an Extended Absence
greeting is in place upon entering the mailbox and asked whether to turn the
greeting off.
¾ Optional Greeting 1 and Optional Greeting 2 – Users can use these optional
greetings to administer a call type, also known as Call Handling. With the Call
Handling feature, users can administer the system to differentiate between
calls that reach a mailbox because an extension is busy or because there is
no answer. Users can decide which optional greeting to activate for Busy calls
and No Answer calls.
In addition from either the graphical user interface or the telephone user interface,
users can record a “Please Hold” prompt as well as their names.
System Failure Message
In the event of a messaging system failure, the proposed Avaya S8720 Media
Server powered by Avaya Communication Manager can be administered to provide
an announcement to callers who would normally be directed to the voice mail
system. The only hardware necessary to provide this feature, an announcement
circuit pack, has been included in the proposed design and pricing.
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Sharing Extensions
A caller application can be set up that allows up to nine users who share a
telephone extension to receive their own voice messages in individual mailboxes.
Callers choose from a single digit menu that plays the list of users associated with
that extension. They then hear that user’s mailbox greeting and can leave a
message. Users then access their individual password-protected mailboxes to
retrieve their messages.
7.2.5 DTMF Signaling
The system will be capable of receiving and generating standard DTMF tone signaling.
Vendor Response Requirement
Confirm support for this feature.
Avaya Response:
Comply.
7.2.6 Greeting
Voice mail calls will be answered on the first ring and be time- and date-stamped.
Vendor Response Requirement
Confirm support for this feature.
Avaya Response:
Comply.
7.2.7 Escape
A caller reaching the voice mail system will have the ability to re-route to an extension by dialing
up to five digits or the operator by dialing "0" before or after leaving a message. It will not be
possible for a caller reconnected to the telephone system to be connected to the public network.
Vendor Response Requirement
Confirm support for this feature.
Avaya Response:
Comply. Callers can choose to enter a different extension or press “0” to reach an
operator. For each voice mail domain, the Administrator can specify all the
conditions for transferring a caller to an operator. A default number and an afterhours number can be set to which callers are transferred when they press “0” to
access assistance. Rotary caller options can be set to either transfer the caller to an
operator or to disconnect after no input is received. For callers experiencing
difficulty navigating the system, a maximum number of errors can be set before
transferring the caller to the operator. The Administrator can also designate a
unique zero-out destination for each user’s mailbox. If a personal operator is not
designated, callers are transferred to a default operator mailbox number.
Measures to prevent toll-fraud are discussed in response to 7.1.1. Human
persuasion and/or misleading statements could make it possible for a caller to be
placed in a position to reach an outside line.
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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.
Avaya Response:
Comply with clarification. Measures to prevent toll-fraud are discussed in response
to 7.1.1. Following these guidelines will reduce the chances of a caller passing
through the attendant to reach an outside line. Human persuasion and/or
misleading statements could make it possible for a caller to be placed in a position
to reach an outside line.
7.2.9 Distribution Lists
The system will contain a minimum of 80 distribution lists of at least 25 names each plus "all
broadcast."
Vendor Response Requirement
Confirm support for this feature.
Avaya Response:
Comply. Personal distribution lists, system distribution lists, and a broadcast to all
feature are all available.
Personal Distribution Lists
A personal distribution list (PDL) is a labeled collection of addresses that subscribers
create and save for later use in order to quickly send a message to a group.
Messages addressed to the mailing list are sent to each list members. Unlike
administrator-created system lists, only the subscriber who created the PDL can
view and use it. A PDL is not available for other subscribers to use. Modular
Messaging supports up to 500 PDLs with up to 999 members (list entries) per
subscriber, subject to system capacity, although no hard limit has been established
for the product. Lists are not subscribers and do not count toward any purchasable
limits.
Subscribers can address messages to PDLs from any of the following interfaces,
applications, or clients:
¾ The Modular Messaging TUIs: Aria, AUDIX, and Serenade for Modular
Messaging
¾ Modular Messaging Outlook Client
¾ Modular Messaging Lotus Notes Client
¾ Modular Messaging Web Client
¾ UCC Speech Access client
¾ A standards-based email client
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System Distribution Lists – Enhanced-List Application
The Enhanced-List Application (ELA) is a messaging tool created and managed by
the Administrator that greatly expands the capability of the Modular Messaging
Message Storage Server to deliver a message to a large number of recipients. The
system supports a maximum of 1,000 ELA lists, with up to 1,500 members per list.
ELA can be nested to create larger lists. Modular Messaging ensures that if a target
recipient is included on multiple lists that are referenced directly or indirectly, the
recipient will receive only one copy of the message.
ELA associates one mailbox to a list of members. When a message is delivered into
the list mailbox, known as the shadow mailbox, the ELA software distributes the
message to the members of the list. ELA members can be local or remote
subscribers, and arbitrary email addresses, thus providing extreme flexibility.
Ordinarily, recipients receive the message as if it were sent by the message
originator. Recipients can reply to the person who originally sent the message and
to all recipients of the original message. However, administrators can configure lists
to block recipients from replying to ELA senders or recipient lists by placing the list
in a communication that cannot receive messages.
An ELA mailbox is just like any other mailbox, allowing such operations as recording
a name for the list, a greeting for the list, and allowing Call Answer messages to be
distributed via ELA. Like any other mailbox, an ELA mailbox has a mailbox number,
a numeric address, and a Modular Messaging email address. The actual addressing
format used to send the message to the list mailbox can be any addressing format
that the relevant interface (TUI or computer interface) supports. The Administrator
can also administer a remote mailbox with the email address of a list, thus making
Microsoft Exchange and IBM Lotus Domino lists available through Modular
Messaging Message Storage Server configurations.
Broadcast
The Broadcast feature enables an Administrator to designate any Enhanced-List
Application (ELA) to be a local broadcast list. When a message is received into an
appropriately configured enhanced-list mailbox, the message is sent to all local
subscribers and to all list members. The Administrator can also set up system-wide
or enterprise-wide broadcast lists. Settings control which subscribers have message
addressing privileges to ELA broadcast mailing lists.
The actual addressing format used to send a broadcast message can be any
addressing format that the relevant interface (telephone user interface or PC
interface) supports. While a new broadcast message does not activate Message
Waiting Indication (MWI), Find Me, and Notify Me, it does activate Call Me, provided
the broadcast message meets subscriber-specified criteria. If customers require
MWI activation, the Administrator can create an ELA list with entries for each local
subscriber in order to activate MWI.
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The Modular Messaging telephone user interfaces announce broadcast messages as
such. They present new broadcast messages before other messages and provide a
summary count of broadcast messages after logging in. The Modular Messaging
Web Client, Modular Messaging Outlook Client, and Modular Messaging Lotus Notes
Client (all graphical user interfaces) provide visual indications for identifying
broadcast messages.
7.2.10 Message Forwarding
Messages may be forwarded to single or multiple destinations with or without introductory
comments.
Vendor Response Requirement
Confirm support for this feature.
Avaya Response:
Comply.
7.2.11 Audit Trail
It will be possible for a user to designate a necessary written record of message destination, input
time and receipt. This audit trail will be printed on the administrative console together with daily
reports.
Vendor Response Requirement
Confirm support for this feature.
Avaya Response:
Comply. A record of messages will appear in the user’s Sent folder with time and
date information. The Transaction Log Database provides a historical record of all
activity in a mailbox. The Administrator can choose to print the log at any time. By
adding the optional IVIZE for Modular Messaging reporting application, this audit
trail can be emailed as an Excel spreadsheet, viewed by the Administrator, or
automatically printed daily.
7.2.12 Message Indication
The receipt of a message in a mailbox will cause a message-waiting lamp or "stutter" dial tone
upon lifting of the station handset to indicate a message-waiting condition.
Vendor Response Requirement
Confirm support for this feature.
Avaya Response:
Comply.
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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.
Avaya Response:
Comply. Please see the response to 7.1.1 for details regarding personal mailbox
security and the use of identification codes (passwords).
7.2.14 Message Recovery
The mailbox owner accessing the mailbox will be automatically told how many new messages
have been received since last access and how many saved messages exist. Upon accessing the
messages, the subscriber will have the choice of deleting, skipping or saving a message. Saved
messages may only be deleted by the subscriber or by the system administrator.
Vendor Response Requirement
Confirm support for this feature.
Avaya Response:
The Modular Messaging telephone user interface provides the number of new
broadcast messages, the number of new voice and fax messages, as well as the
total number of saved messages stored.
From the graphical user interfaces, the subscriber can easily prioritize message
review. Subscribers can also mark read messages as unread, thus moving
messages from the Saved category to the New category.
Using either telephone or graphical user interfaces, subscribers can choose to save,
skip, or delete messages, as well as forward or reply. Saved messages may only be
deleted by the subscriber or the system administrator.
7.2.15 Message Reply
A mailbox owner may respond to a message input by another system mailbox owner by simply
depressing a single key.
Vendor Response Requirement
Confirm support for this feature.
Avaya Response:
Comply. After listening to a voice message, users press “8” to access the following
reply options. Replies can be made with or without the original message attached,
and can be made to only the message sender or all message recipients (which can
be listened to before choosing to reply to all). If supported by integration, users can
also call the sender.
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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.
Avaya Response:
Comply.
7.2.17 User Controls
A user accessing their mailbox will be capable of the following control functions:
1.
Playback messages
2.
Skip to next message
3.
Cancel review
4.
Replay last message
5.
Replay faster or slower
6.
Pause
7.
Append information
8.
Forward message (to mailbox or list)
9.
Create new answer announcement
10.
Increase play-back volume
Vendor Response Requirement
Confirm support for this feature. Indicate if any function is not supported.
Avaya Response:
Comply. The proposed solution supports all of the above features.
7.2.18 System Management Console
The system will be equipped with a CRT and printer to provide system management functions.
The administrative programs and traffic information secured will be possible during system
operation. Traffic reports will be available on customer demand or automatically on a preprogrammed 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 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.
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Avaya Response:
Does not comply with clarification; Modular Messaging provides detailed standard
reports. Once generated, reports can be viewed onscreen and automatically sent to
a printer, if desired; however, the printer is not provided as part of the solution. In
addition, reports can be exported for archiving or additional manipulation purposes
by using alternative tools. The export facility supports a number of popular
spreadsheet, word processor, and data interchange formats. These include
character separated values, comma separated values, Crystal Reports, data
interchange format, Microsoft Excel, Lotus 1-2-3, record style (columns of values),
Rich Text Format, tab-separated text, tab-separated values, ASCII text, and Word
for Windows. An exported report file can also be attached to a message sent via a
MAPI-enabled email system.
The Message Storage Server is a Linux server; as such, separate storage indicators
are not provided or needed to indicate the space utilized by announcement,
information mailboxes, and messages. Linux server reports can be utilized to see
the amount of overall storage used.
Following are the available standard reports for each server.
Avaya Messaging Application Server Reports
The Reporting Tool can be used by Administrators to generate predefined
Messaging Application Server Reports for monitoring voice mail system usage,
planning capacity, and tracking system security. The information is taken from the
Transaction Database and generated for the voice mail domain. Some reports can
also generate Messaging Application Server-specific or subscriber-specific
information.
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Using the Reporting Tool, Administrators can generate the following Messaging
Application Server standard reports.
Hourly
Statistics
Records detailed information regarding the number of incoming
and outgoing calls each hour in a specified time period. This
information is useful for monitoring incoming and outgoing call
patterns for the voice mail domain.
Login Failures
Records detailed information regarding unsuccessful mailbox
logins due to an incorrect password or invalid mailbox number.
A defined time period can be specified. This information is useful
to help monitor system security for the voice mail domain.
Basic Metric
Records statistics on messaging activity in the voice mail
domain. It includes general information on telephone user
interface usage and statistical information on subscriber
telephone user interface logins. The contents of the Basic
Metrics report are the core Key Performance Indicators for the
voicemail system; hence this report also provides a general
performance overview.
Port Statistics
Records detailed incoming and outgoing call information for
each port configured in the voice mail domain. The time period,
ports, and MASs can be specified. This information is useful for
monitoring port usage.
System Usage
Records detailed call and messaging statistics for the voice mail
domain. These include statistics regarding general call
information, callers’ actions, incoming and outgoing call
summaries, and message summaries. A defined time period can
be specified. This report can be used to help monitor usage of
the system for such things as the types of calls, the number of
messages left, the number of fax calls, and the time ports were
used for text-to-speech, and their disposition.
User Mailbox
Statistics
Records detailed information regarding telephone calls and
messages received by each mailbox in the voice mail domain.
The time period and mailbox can be specified.
Avaya Message Storage Server Reports
The Avaya Message Storage Server collects information about system settings and
attributes and information that depicts how the system is used, including data
about features, subscribers, communities, data port loads, and remote messaging
traffic. This information is displayed in real-time dynamic report pages, and in
messaging traffic reports.
These reports reflect:
¾ Thick Client Access (Microsoft Outlook, Lotus Notes)
¾ Web Client Access
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¾ UCC Speech Access, Access
¾ Message Networking Access
REPORT
DESCRIPTION
Community
Daily or Hourly
Traffic Report
These reports show the total number of messages sent and received
by each community. They also show the number of messages that
were not sent or received by each community due to restrictions on
sending during any day in the last 32-day period or any hour in the
last seven days.
Feature Daily or
Hourly Traffic
Report
These reports show traffic information on a feature-by-feature basis.
Features are divided into call answer features and messaging features.
Load Daily or
Hourly Traffic
Report
These reports show daily load traffic information for 1 to 32 days or
hourly traffic information for the last 7 days. Traffic load refers to the
message traffic and storage relative to established mailbox thresholds.
Network Load
Daily or Hourly
Traffic Report
These reports show network channel traffic one day at a time for up to
32 days or one hour at a time for any hours within the last 7 days.
These reports can show any nodes that are exceeding specified
threshold limits, the number of calls that went unanswered, the
number of calls on each channel, and other channel traffic
information.
Remote
Message Daily or
Monthly Traffic
Report
These reports show information about traffic loads between a local
messaging machine and a specified remote messaging machine.
Report of
Classes of
Service
This report shows information about the current classes-of-service.
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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
Avaya Response:
Does not comply with clarification; Please see the response to 7.2.18 for reports
that will provide numbers 2 through 5. Number 6 will be provided by the S8720
Media Server. Reports cannot be scheduled in incremental hours as requested.
7.2.20 System Changeability
It will be possible for the system administrator to add and/or delete mailboxes, change general
recordings and perform other administrative duties while the system is in operation.
Vendor Response Requirement
Confirm support for this feature
Avaya Response:
Comply.
7.3.0 Networking
VoiceCon plans on networking it new HQ messaging system to other VoiceCon locations
equipped with messaging systems.
Avaya Response:
Read and understood.
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7.3.1 AMIS
The proposed messaging system should support AMIS networking standards.
Vendor Response Requirement
Confirm support for these features
Avaya Response:
Comply via the optional Avaya Message Networking system. Modular Messaging
uses Internet standard networking protocols for message transport and directory
updates. It supports the SMTP/MIME protocol for message transport and supports a
Lightweight Directory Access Protocol (LDAP)-based protocol to communicate with
other networked Modular Messaging systems. When networked with an optional
Avaya Message Networking system, Modular Messaging servers can communicate
via such protocols as AMIA-A and VPIM to other messaging servers that use those
industry standard protocols, as well as to servers using Avaya proprietary network
protocols.
7.3.2 Digital IP Networking
The proposed messaging system should support VPIM networking standards.
Vendor Response Requirement
Briefly describe digital networking capabilities of your proposed messaging system solution.
Indicate if VPIM is supported.
Avaya Response:
Comply via the optional Avaya Message Networking system. The optional Avaya™
Message Networking system is a network integrator, allowing Modular Messaging
Servers to communicate with other messaging servers that use supported industry
standard and Avaya proprietary network protocols. This turnkey server-based
solution connects individual voice or multimedia messaging systems. Using a
"store-and-forward" approach to networking, the Message Networking solution
receives messages, performs the necessary protocol conversions, and delivers the
message to one or multiple recipients on one or multiple messaging systems. The
hub-and-spoke network topology, where each messaging system is directly linked
through the Message Networking solution to every other messaging system, is
simpler to manage and easier to expand than a conventional point-to-point
topology.
Each messaging system requires only a single connection to the Message
Networking solution, and new systems can be added easily without affecting
existing servers. All message routing, protocol conversions, administrative
functions, and management capabilities reside on the Message Networking solution,
so network connections are less costly and easier to manage.
The Message Networking solution eliminates long-distance charges when customers
use digital TCP/IP to transport messages over their wide area network (WAN). In
addition, the Message Networking solution can network from 2 to 500 remote
locations. It can expand messaging capabilities as customers' networks grow.
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Message Networking provides transcoding between diverse digital voice encoding
formats and converts analog-to-digital and digital-to-analog. The Message
Networking solution supports TCP/IP (for INTUITY™ AUDIX®, Octel® 200/300,
Octel® 250/350, and non-Avaya messaging servers that are VPIM V2-compatible),
AMIS-analog, and Octel-analog protocols. To design and implement the Avaya
Message Networking solution, customers have the option to administer their own
end points (thereby reducing their ongoing implementation costs) or choose one of
the specially designed Professional Services offers. With one of these offers, the
Professional Services Organization designs and/or implements the Message
Networking for our customers. Please note that if your network includes
Interchange as one of the end-nodes, a Professional Services offer is required to
implement the Message Networking solution.
Through various methodologies, the Avaya Message Networking server maintains a
complete and current enterprise-wide directory of all users in the network. Up to
500,000 networked users are supported. The Message Network server can provide
all or selected views of the enterprise directory to the Modular Messaging server,
complete with ASCII names and network addresses. The directory updates can also
include up to 120,000 spoken names. These directory updates ensure that users
can address by name and receive name confirmation while addressing messages to
networked users.
In addition to the above-mentioned features, the Message Networking solution also
offers:
¾
LDAP-based Directory Update Support for a Modular Messaging solution
¾
Web-based Administration Interface
¾
Test Connect Tools
¾
On-line Help
¾
Call Detail Recording
7.4
Integrated Messaging Application
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 textto-speech conversion.
Avaya Response:
Avaya is proposing an Avaya Modular Messaging solution with an Avaya Message
Storage Server. To provide unified messaging capabilities, graphical user interface
(GUI) clients can be made available to the user to provide access to voice, fax, and
optionally email messages from a PC. These applications provide a visual interface
to perform various operations such as accessing and sending messages; managing
messages; configuring mailboxes; and maintaining mailboxes, rules, greetings, and
TUI and GUI preferences.
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The following GUIs apply to Modular Messaging with a Message Storage Server.
¾ Avaya Modular Messaging Microsoft Outlook Client: This client application
creates a new Inbox for Modular Messaging messages, separate from the
corporate email inbox. Users can access, send, and manage messages from
this Inbox within the Microsoft Outlook email application.
¾ Avaya Modular Messaging IBM Lotus Notes Client: This client application
creates a new Inbox for Modular Messaging messages, separate from the
corporate email inbox. Users can access, send, and manage messages from
this Inbox within the IBM Lotus Notes email application.
¾ Avaya Modular Messaging Web Client: This Avaya client application allows
users to access, send, and manage voice, text, fax, and corporate email
messages from a Web browser. While it cannot be used for mailbox
administration functions, users can still do this from the TUIs or from
Subscriber Options.
¾ Standards-based Clients: Users can receive, send, and manage messages
from a standards-based email client that supports either IMAP4 or POP3,
although IMAP4 is preferred for its enhanced capabilities. While this method
cannot be used for mailbox administration functions, users can still do
mailbox administration from the TUIs or from Subscriber Options.
¾ Subscriber Options: This application provides users the capability to modify
their mailbox settings from a desktop PC. Subscriber Options integrates with
the Microsoft Outlook or the IBM Lotus Notes email client. It can also work as
a stand-alone application.
¾ Web Subscriber Options: This application provides users the capability to
modify their mailbox settings from a Web browser. Web Subscriber Options
provides all of the functionality that Subscriber Options provides. Authentic
users must enter their mailbox numbers and numeric passwords to enter
Web Subscriber Options. Users can also log in to this application using a
Quick Logon from the Modular Messaging Web Client.
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The solution also includes Avaya UCC Speech Access, which provides the following
capabilities via integration with Outlook.
Access to & Control of the information already in place!
Messages
•
•
•
•
•
•
Read
Reply
Forward
List
Filter
Send to a
contact
Contacts
• Call a contact
• Send messages to a
contact
• Get phone numbers
• Get email addresses
• Get mailing address
Calendars
Calling
• Review Appointments
• Create Appointments
• Read, reply to and forward
meeting requests
• Accept or decline meeting
requests
• Set Reminders
• Call any contact
• Dial any number
• Reply to voice
message with a call
• “Single Number”
Find-me/Follow-me
(Hide-me)
Tasks
•
•
•
•
Read
List
Filter
Reply to
request
44
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©2006 Avaya Inc.
Conference Calls
• Add up to 23 parties to
conference
• Speak commands during
conference
• Leave and re-enter conference
• Continue two-party call after
conference
Avaya – Proprietary Use pursuant to Company instructions
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Part 1 – Section 8 - Contact Center
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.
Avaya Response:
Read and Understood.
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.
Avaya Response:
Comply. Avaya Call Center Basic is included with each and every Avaya
Communication Manager 3.1 system without requiring a separate ACD system,
server, hardware, or software. Call Center Basic includes basic ACD functionality
including call queuing, basic announcements, direct (linear) call distribution or
uniform call distribution (most idle agent hunting), agent login/logout, agent work
states (Auxiliary Work, After Call Work, Auto In, Manual In, Auto Available, Auto
Answer), agents logged into multiple splits, multiple call handling, Service
Observing, VuStats, and more. And as of Communication Manager 2.1 (July 2004),
Basic Call Management System (BCMS) is now provided with Basic Call Center at no
additional charge. BCMS provides “built-in” Call Center reporting capabilities
without requiring any adjunct server or software.
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Part 1 – Section 8 - Contact Center
For more advanced ACD requirements, Avaya Call Center 3.1 is now available, as
an optional overlay to the included Call Center Basic, in three (3) scaleable
packages designed to suit the particular needs of your business:
1. Avaya Call Center Introductory Offer: Provides all of the ACD
functionality required to operate a call center with full vectoring basic and
advanced routing capabilities. It now can handle up to 50 agents at an
extremely attractive per agent price.
2. Avaya Call Center Elite: Our most popular package features Avaya Expert
Agent Selection (EAS) skills-based routing and the full complement of
advanced
Call
Vectoring
conditional
routing
capabilities—customer
programmable call screening, analysis of ANI/DNIS, trunk group, and call
prompted digits; customer programmable call routing, including caller
selected routing, agent specific routing, customer programmable call queuing,
including
multi-split
queuing
and
intelligent
overflow;
customer
programmable call treatments, including customized and estimated wait time
announcements, and hold in queue; and basic call data collection and
reporting, including system, group and agent level reports. In addition, the
Elite package now includes:
Avaya Virtual Routing – offers the ability to have separate Call Centers
operate as a single virtual Call Center by effectively load balancing work
activity across centers.
IP Agent Shared Control – provides a common soft phone desktop for all
agents and allows all agents to be IP enabled. IP Agent shared control allows
the IP Agent application to control DCP or IP terminals directly, i.e. without
the need for CTI support. Agents can use either the IP Agent application or
the telephone phone buttons for feature activation or call control. It is used
for an on-site agent option to enhance the capabilities of the DCP or IP
terminal through the use of the IP Agent user interface. This now allows for a
consistent agent soft phone desktop for all agents, enabling even legacy DCP
CallMaster terminals to function as IP Agents. IP Agent Shared Control
licenses enable advanced communication application features such as instant
messaging, screen pop and VuStats Monitor, all of which can provide agent
productivity enhancements.
3. Avaya Call Center Elite Enterprise Edition: Adds Avaya Business
Advocate and Avaya Advanced Segmentation to the standard Elite package
at a very attractive price point.
The following table highlights the Avaya Call Center features available with each
Call Center package. Communication Manager provides Call Center Basic as a
standard part of the software package. Call Center 3.0/3.1 enhancements are
indicated by a *.
December 1, 2006
©2006 Avaya Inc.
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Avaya Call Center Software
Automatic Call Distribution (ACD)
Auto Available Split (AAS)
Dial Plan Expansion to 7-Digits
Increased Logged-in Agent-Split Pairs (see note 6)
Increased Number of Announcements (see note 6)
Measured Trunk/Trunk Groups Increase (see note 6)
Most Idle Agent (MIA) Across Splits/Skills Option
MIA Treatment for After Call Work (ACW)
Multiple Call Handling - Requested
Multiple Call Handling - Forced
Move Agent/Change Splits-Skills While Agent Staffed
Multiple Announcement Boards (see note 7)
Redirection On No Answer (RONA)
Remote Logout of Agents
Service Observing – Basic
Service Observing by Class of Restriction (COR)
Service Observing - Remote
Timed After Call Work (TACW) / Agent Pause Between Calls
VuStats, including enhancements:
- Service Level
- Login IDs
Call Vectoring – Prompting (included with CM basic as AutoAttendant). Includes: - Administrable Inter-Digit Timeout
Holiday Vectoring (included with CM basic for use with Attendant
Vectoring) (also see note 1)
BCMS (Basic Call Management System - see note 2)
Dynamic Hunt Group Queue Slot Allocation
Redirection on IP Failure
Call Work Codes (CWC)
Call Vectoring - Basic
Redirection On No Answer to VDN (Vector Directory Number)
Support for Network Provided Digits (CINFO - see note 2)
Service Observing of VDNs (Vector Directory Numbers)
Service Observing Upon Agent Answer
Display VDN for Route-to Direct Agent Call
VDN of Origin Announcement (VOA)
VDN Return Destination
Vector Administration:
- Route-to with or without coverage
- Support for Multiple Audio/Music Sources On Delay
Vector Initiated Service Observing
December 1, 2006
©2006 Avaya Inc.
Avaya Call Center 3.1
Package
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes*
Yes
Yes
No
No
No
No
No
No
No
No
No
No
No
No
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
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Part 1 – Section 8 - Contact Center
Avaya Call Center Software
Vectoring Advanced Routing:
- ANI (Automatic Number Identification)/II (Info Indicator) Digits Routing
- ASA (Average Speed of Answer) Routing
- EWT (Expected Wait Time) Routing
- VDN Calls Routing
Best Service Routing (BSR) Single Site
BSR Available Agent Adjust-By
BSR Available Agent Wait-Improved
BSR Polling Over IP without B (Bearer) Channel
BSR Local Feedback for IP and ISDN Queued Calls*
Duplicate Agent Login ID Administration
Voice Response Unit Connect & Disconnect (C & D) Tones (DTMF Signals)
Least Occupied Agent (can replace Most Idle Agent with either UCD or EAD)
Expert Agent Selection (EAS):
- Add & Remove Skills using FAC (Facility Access Codes)
- Agent Call Handling Preference (ability to specify)
- Auto-Answer/Manual-Answer Specified by EAS Logical Agent
- INSPECT Button shows station name
- MWL (Message Waiting Lamp) for Logical Agent Coverage
- Service Observe – Logical Agent
- Increased Capacities [up to 60 skills per agent and 16 skill level preferences
using EAS-PHD (Preference Handling Distribution)]
Reason Codes:
- AUX (Auxiliary) Work Reason Codes(see note 3)
- Logout Reason Codes
Increased Login IDs to 20,000 (see note 6)
Increased Administered Login ID-Skill Pairs to 180K (see note 6)
Increased Skill (Hunt Groups) to 2,000 (see note 6)
Network Call Redirection
Service Level Maximizer
Maximum Agent Occupancy
Variables in Vectors (see note 4)
Advanced Segmentation* (see note 5)
Vectoring Enhancements* – VDN Variables; Subroutines, etc.
Location Preference Distribution*
ACD Options by Agent*
Forced Agent Logout from After Call Work*
Virtual Routing (see note 8)
IP Agent Shared Control (see note 8)
December 1, 2006
©2006 Avaya Inc.
Avaya Call Center 3.1
Package
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
Yes
Yes
Yes
Yes
Yes
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
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Part 1 – Section 8 - Contact Center
As of Communication Manager 2.1 (July 2004), BCMS is now provided
with Basic Call Center at no additional charge.
There is also an Enterprise Edition package that includes Business
Advocate and Advanced Segmentation with Elite at a very attractive price
point
Notes:
Note 1: Holiday tables have been increased to 100 with 3.0. The additional 90
tables are only available to Elite customers.
Note 2: BCMS internally measured agents increased from 2,000 to 3,000 in 3.0 call
center.
Note 3: Aux Reason Codes can be increased to 100 with Call Center 3.0.
Note 4: Variables in Vectors was introduced in 2.0 with enhancements made in 2.1
and 3.0.
Note 5: Advanced Segmentation was introduced in 2.1 with enhancements made in
3.0 to provide a simple screen-pop capability with IP Agent R6.
Note 6: The increased capacities are only available with the Avaya S8700 or S8500
Server platforms (Multi Connected or IP Connected configurations).
Note 7: Multiple announcement boards/sources (TN2501s or embedded virtual VAL
sources integrated in Media Gateways/Servers) are supported even without
a Call Center package administered.
Note 8:
New Call Center packaging available in November 2005 now provides
Avaya Virtual Routing and IP Agent Shared Control as entitlements of Call
Center Elite for new 3.0 Call Centers.
Hardware & Software Requirements
The ACD functionality for the voice channel is provided by Avaya Communication
Manager Release 3.1 running on the Avaya S8720 Media Server directly without
requiring a separate, adjunct ACD hardware platform.
December 1, 2006
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Part 1 – Section 8 - Contact Center
Capacities
Avaya Communication Manager 3.1 ACD
Capacities
S8720 IP Connect
AUTOMATIC CALL DISTRIBUTION (ACD)
Announcements per Split
Announcements per System
Splits
ACD Members per Split
Max. Administered ACD members4.4
Logged-In Splits per Agent5
Max. logged-in ACD agents (per system) when each logs into:
1 Split
2 Splits
3 Splits
4 Splits
Queue Slots per Group7
Queue Slots per System7
CALL VECTORING
Skills a Call Can Simultaneously Queue to
Priority Levels
Recorded Announcements / Audio Sources for Vector
Delay
Vector Steps per Vector
Vector Directory Numbers
Vectors per System
Simultaneous Active Subroutine Calls
Number of Collected Digits for Call Prompting or CINFO
Number of Dial-Ahead Digits for Call Prompting
Vector Routing Tables (100 entries per table)
BSR Application Routing Tables (forms)
BSR Application-Location Pairs20.5
Holiday Tables (15 entries per table)
EXPERT AGENT SELECTION (EAS)
Skill Groups
VDN Skill Preferences
Max. Skills a Call Can Simultaneously Queue to
Max. Administered ACD Members (login ID / Agent-Skill
pairs)28.1
Max. Staffed (logged-in) ACD Members28.3 i.e.., agent-skill
pairs
Max. Administered Agent Login IDs28.4
Max. Skills per Agent
R13 CMS
Skill Levels (preferences) per Agent Skill
Max. Staffed (logged-in) EAS Agents per Skill (members
per group)
December 1, 2006
©2006 Avaya Inc.
6
2
3,000
2,000
1,500
60,000
4
5,200
5,200
5,200
5,200
NA
NA
3
4
3,000
32
20,000
999
8,000
16
24
100
511
2560
99
2,000
3
3
180,000
60,000
20,000
60
16
3,000
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Part 1 – Section 8 - Contact Center
Avaya Communication Manager 3.1 ACD
Capacities
Max. Logged in EAS Agents (per system) When Each Has:
1 Skill
2 Skills
4 Skills
10 Skills
20 Skills
60 Skills
S8720 IP Connect
6
5,200
5,200
5,200
5,200
3,000
1,000
Note: The capacity limits for System and Per Group Queue Slots are not applicable
with any platforms that run Avaya Communication Manager Release 2.1 or later
due to the Release 2.1 Dynamic Hunt Group Queue Slot Allocation feature. The
Dynamic Hunt Group Queue Slot Allocation enhancement available with Avaya
Communication Manager 2.1 will now allocate dynamically Queue Slots on a shared
basis across the entire media server driven system as needed and when needed. It
reduces the need to administer queue slots on each Hunt Group form. If queue
limiting is required, then vector conditionals can be used. Hunt group queue slots
are now allocated on an as needed basis allowing all calls that are possible to be in
queue. The common pool of queue slots is 12,000 for the S8500/S8720 platforms.
Please refer to the Avaya Communication Manager Release 3.1 System Capacities
for applicable notes.
Announcement capacities and maximum recording time are dependent upon the
type of announcement hardware and the system platform. The various types of
announcement hardware and their associated capacity per system platform are
shown in the table below. Please refer to the Avaya Communication Manager
Release 3.1 System Capacities for applicable notes.
Avaya Communication Manager 3.1 Announcement
Capacities
S8720 Media Server
Announcement/Audio Sources per System 18
3,000
Analog & Aux Trunk Announcements
Queue Slots per Announcement
1,000
Queue Slots per System
1,000
Calls Connected to Same Announcement
1,000
Integrated Announcements
Queue Slots for System
4,000
Calls Connected to Same Announcement
1,000
TN2501 Circuit Packs (VAL)
10
Total Announcement Sources: Integrated Circuit Packs on G650s
10 TN2501 + 250 vVAL
(up to 10) plus embedded vVAL Sources on G250/G350/700 MGs
TN2501AP (VAL) Circuit Packs on G650 Port Network Gateways
Channels per Circuit Pack (Playback Ports)
31
Maximum Announcements per Circuit Pack
256*
Circuit Pack Content Saved
All active Circuit Packs
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Part 1 – Section 8 - Contact Center
Avaya Communication Manager 3.1 Announcement
Capacities
S8720 Media Server
Recording Time per Circuit Pack (in Minutes) 90
Low-end Option (Max. 1 Circuit Pack)
10
High-end Option (with up to 5 Circuit Packs for CSI; 10 for
60
S8700 Series)
G650 Embedded Integrated SSP (Scalable Speech Processor) Announcements
SSP Circuit Pack
1 per G600
Channels per SSP Integ. Annc. Circuit Pack
8
Maximum Announcements per Circuit Pack
128
Circuit Pack Contents Saved
All
Recording Time (Min)
16 KB recording
240
32KB recording
120
64KB recording
60
Embedded Media Gateway Integrated Virtual VAL (Voice Annc. Over LAN) vVAL
Announcement Sources
Channels per Source (playback ports) - depends on the Media
6 or 15
Gateway (lower number for G250/G350, higher number for
G700)
Maximum Announcements per Source
256
Source Contents Saved (VAL FTP download)
All active Circuit Packs
Recording Time per Source in Minutes - depends on the Media
10 or 20
Gateway (lower number for G250/G350, higher number for
G700)
LOCALLY SOURCED MUSIC and ANNOUNCEMENTS (LSMA) - Allows
"announcements" & "audio groups" to also be used as Music on Hold sources.
Audio Groups (for announcements/music)121
50
Sources per Audio Group (VAL and/or vVAL)
250
Announcement extension-source combinations122
3000
MOH Groups (for assignment as the system music source or
10
Tenant Partition Multiple Music Source)
Analog/Aux Trunk Sources (Ports) per MOH Group
250
Unique Analog/Aux Trunk MOH Ports per System (each
250
referenced only once)
December 1, 2006
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Part 1 – Section 8 - Contact Center
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.
Avaya Response:
ANI Routing
Comply. There are four call vectoring features that allow routing by ANI if received
from the network facilities:
ANI Routing Directly Within the Call Vector – ANI can be compared to numbers
programmed directly within the call vector step. Some ANI examples are shown
below:
goto step XX if ANI = none
if no ANI is provided
goto step XX if ANI = 3038460064
if ANI matches a number
goto step XX if ANI <= 9999999 if ANI is less than 7 digits
goto vector XX if ANI = 212+
if ANI is from a particular area code
goto vector XX if ANI <> 212841+
if ANI is not from a particular office
ANI Routing Using Call Vector Routing Tables – Vector Routing Tables contain a list
of numbers that can be used to test a Goto if Digits/ANI command. The values can
be tested to see if the ANI or prompted digits are or are not in the specified table.
Some ANI examples using Vector Routing Tables are shown below:
goto vector XX if ANI in table 6
if ANI is listed in table 6
goto vector XX if ANI not-in table 7
if ANI is not listed in table 7
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Part 1 – Section 8 - Contact Center
ANI Routing Using Converse Vector Command – ANI can also be passed by the
converse vector command to an adjunct such as the Avaya Interactive Response
(IR) system for larger groups of numbers or if additional information is required
before routing the caller to an agent. Custom Call Routing applications for the
Avaya IR can perform database lookups based upon the ANI or digits received from
the call vector or interactive voice response scripts can be used to determine the
desired destination. This destination can then be passed back to the Avaya Media
Server or Avaya DEFINITY® Server and used as a route-to destination by the call
vector. Because the digits passed and received are via Inband DTMF signaling, no
special facilities are required.
ANI Routing Using an Adjunct Application – The optional Adjunct/Switch Application
Interface (ASAI) is available to provide routing instructions from an adjunct
application such as: Avaya Interaction Center, Avaya Voice QuickStart, Avaya
Interactive Response, Avaya Contact Center Express, or Avaya Advanced
Segmentation for other CTI routing applications. When a call reaches the adjunctroute vector command, vector processing is suspended while awaiting routing
instructions from the adjunct application. Vector processing will continue (after a
programmable time limit) if no instructions have been received. This enables calls
to be processed in the event of a computer link failure. The adjunct computer can
view system-wide conditions and enterprise databases to determine where to route
the call–to a specific agent, an agent group, a non-ACD user, or any other valid
destination.
DNIS Routing
Comply. With Call Vectoring, DNIS digits received from the network facilities can be
mapped directly to a Vector Directory Number (VDN) extension and an associated
call vector can be provided for each DNIS application. Or a single call vector may
handle multiple DNIS applications. Utilizing DNIS digits to identify applications
allows multiple applications to share trunking facilities. The associated call vector
evaluates all conditions and determines routing, queuing, prioritization, and call
handling treatment based upon specified conditions or adjunct routing instructions
received from a CTI application. The VDN is typically assigned a 16-character name
and serves to identify the type of call to the agent. Thus, agents can handle calls
for multiple applications and be informed via their voice terminal display of the type
of each call as it arrives so that they can answer appropriately.
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Part 1 – Section 8 - Contact Center
Routing Based on Call Volumes
Comply. With Call Vectoring, routing and queue size for each split/skill can be
managed dynamically using vector commands. Thus, the desired routing and queue
size can change based on various conditions in your call center. The following types
of conditions can be checked to determine alternate routing to manage queue
length:
Number of calls queued
Number of connected calls by Vector Directory Number (VDN)
Number of staffed agents
Number of available agents
Expected Wait Time
Rolling Average Speed of Answer
Oldest Call Waiting Time
Time of day, day of week, date of year
For example, you can test for the number of calls queued in a specified split/skill
before queuing a call. If a call is not queued, the call can receive a forced busy
signal, be disconnected (generally after hearing an announcement), or routed to
another split/skill, an individual station user, the attendant console, or voice
messaging. Since a forced busy signal on digital facilities does not return answer
supervision, no billing is incurred on usage-sensitive lines, eliminating unnecessary
use of expensive facilities. This capability can also be used to limit the number of
incoming calls being held in queue for toll free call centers.
Routing and queue size can also be determined by the number of active calls in a
specific application. With Call Vectoring, this is done by Vector Directory Number
(VDN), which is usually the dialed number or the result of a prompt to the caller.
This is an excellent method for applications that require a contracted or
predetermined number of agent resources. When the number of calls exceeds the
desired level, the caller hears a busy signal or announcement providing alternate
choices. This also enables multiple applications to share facilities while preventing
one application from utilizing all of your call center resources.
December 1, 2006
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Part 1 – Section 8 - Contact Center
Routing Based on Performance Criteria
Comply. The Service Level Maximizer (SLM) feature, available as a standard
component of the Avaya Elite Call Center package, allows you to better utilize your
agents to match the needs of your business. You can specify desired service levels
for each queue and have the ACD automatically prioritize and distribute calls in an
order that assists each queue come closest to achieving their desired service
objectives. SLM maximizes agent utilization to meet percent within service level
targets:
Expressed in service level terms such as “answer 80% (X) of skill 1 calls
within 20 (Y) seconds”
Allows different targets to be administered (it’s now just “X”s and “Y”s to
each skill to differentiate service based on the value of this type of call to the
business
Assigned on a per skill (hunt) group basis.
The SLM agent selection method is based on user-defined target service levels for
SLM-administered skills and the concept of agent opportunity costs. SLM provides
an alternative agent selection process that is designed to:
Compare the current service level for each SLM-administered skill to a userdefined call service level target and identify the skills that are most in need
of agent resources to meet their target service level.
Identify available agents and assess their overall opportunity cost, and select
only those agents whose other skills have the least need for their service at
the current time.
Because SLM is able to differentiate skills in terms of their current call service
demands, it provides the following advantages over other agent selection methods:
Since agent resource needs for each skill are assessed in real-time, you can
use SLM to allocate agent resources to those skills that have the greatest call
service demand in a dynamic manner, thereby reducing overall call response
times.
Potential problems associated with staffing exceptions, or fluctuating, intraday call service demands are also reduced.
SLM is especially useful for call center operations that are bound by contract
or other legal obligation to meet specific service level requirements.
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Part 1 – Section 8 - Contact Center
Call Vectoring also supports routing based upon current performance criteria in your
call center. This enables you to maximize your efficiency and use of agent and
network resources and offer Best Service to your callers with the following
conditional routing capabilities:
Rolling Average Speed of Answer for a specified split/skill or Vector Directory
Number (VDN).
Amount of time that the oldest call in a specified split/skill queue has waited
to be answered.
Current Expected Wait Time for a specified split/skill or for the best identified
split/skill.
Current Expected Wait Time for the call being processed.
Current Expected Wait Time or Adjusted Wait Time for a specified split, skill,
or location being considered (with optional MultiSite Best Service Routing).
Predicted Amount of Improvement in Expected Wait Time for split/skill.
Current Queue Position for Interflow.
All of this checking can be performed prior to queuing a call, or at anytime
subsequent to queuing the call. Multiple split/skill queues can be checked. The call
can be simultaneously queued to up to three different split/skill groups and be
answered by the first available agent in any of the groups. The call can be
automatically queued to backup split/skills or queued conditionally based upon
defined overflow conditions.
Priority Queuing
Comply. Selection of which call to deliver to an available agent is influenced not
only based upon priority of the calls in queue (described below) but also based
upon agent skill levels, call handling preferences, and routing algorithms such as
Service Level Maximizer or optional Business Advocate. Avaya has the ability to
tailor a sophisticated call prioritization scheme to help you meet your specific
business goals.
For basic call prioritization, Call Vectoring offers four levels of entry to an ACD
queue including:
Low priority
Medium priority
High priority
Top priority
Using these four levels, preferential answering treatment can be given to certain
incoming calls based on various criteria. These criteria might include the cost of
various trunking facilities, the amount of revenue generated by certain calls, and
special courtesy to customer groups or executive personnel.
December 1, 2006
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Part 1 – Section 8 - Contact Center
Priority can be assigned at the incoming trunk group, by dialed number, or by caller
prompted information. Priority can also be changed on a dynamic basis, according
to current conditions such as: time call has been in queue, number of calls, number
of agents available, number of agents staffed, time of day, and/or day of week.
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.
Avaya Response:
Agent Skills
Comply. Expert Agent Selection (EAS) is available with Call Center Elite. EAS routes
incoming Automatic Call Distribution (ACD) calls to the agent who is best qualified
to handle the call, that is, the agent with the specialized skills or experience
required to best meet the caller’s needs. With Expert Agent Selection Preference
Handling Distribution (EAS-PHD) on an Avaya S8720 Media Server, agents may be
assigned up to 60 skills each at one of 16 levels of preference or proficiency subject
to system maximums for active agent/skill pairs.
Customer Preference
Comply. You can easily accommodate customer preferences within your routing
specifications by using Avaya Integrated Call Prompting, a standard feature of Call
Center Elite. Call Vectoring provides voice response, or Call Prompting, capabilities
using standard Recorded Announcements along with Digit Collection features. This
functionality is a part of the Avaya Call Center integrated hardware and software
feature set and does not require any peripheral system.
Call Prompting can be used to accommodate customer routing preferences in the
following ways:
Provide Auto Attendant functionality; for example: “If you know the 5 digit
extension of the party you wish to speak with, you may enter it now…”
Provide an Auto Attendant menu of routing options such as “Press 1 for
sales, Press 2 for service, press 3 for billing…”
Provide caller with messaging options such as “Your Estimated Wait Time is
3-4 minutes, if you would prefer to leave a voice mail message and have an
agent call you back, please press 1…, otherwise, please continue to hold…”
Provide vector routing table or database assisted routing options such as
“Please enter your 10 digit account code…” or “Please enter your zip code…”
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Part 1 – Section 8 - Contact Center
Preferred Customers
Comply. You can recognize your preferred customers based upon ANI or caller
prompted information such as account numbers, etc. and provide appropriate call
treatment and prioritization. For example, important customers may have a
dedicated agent who handles their calls when possible.
Inbound and Outbound Call Levels
Comply. Call Vectoring can analyze, route, and queue based upon current call
volumes.
For an Inbound and Outbound Blended solution, the optional Avaya Proactive
Contact System is recommended. The Avaya Proactive Contact System allows
agents to reach more customers, more quickly and more profitably. Whether a
calling mission requires inbound, outbound or blended solutions, the Avaya
Proactive Contact System provides unparalleled technology to meet the demands of
every customer's business. As inbound volume increases, our sophisticated callblending applications smoothly transfers available calls to the blended inbound or
outbound team as needed. You can choose from two blending strategies: blending
based on either overflow or on predictive analysis of inbound call trends. Sporadic
inbound overloads and agent idle time are minimized, while contact center
productivity is maximized.
Multi Media
Comply. For larger multimedia contact center requirements, we recommend the
Avaya Interaction Center.
For your mid-sized contact center, the new Avaya Contact Center Express will easily
support your 120 agents at a cost-effective price. Contact Center Express provides
robust multi-channel routing capabilities designed for contact centers with 50-150
concurrent users/agents who want to gain efficiency by implementing multiple
channels of customer communication (voice, email, web) and Application
Enablement Services (AE Services) capabilities such as Screen-Pop, Segmentation
Routing or Interactive Voice Response (IVR) integration.
Avaya Contact Center Express manages the collection, queuing, and delivery of
voice and non-voice work items such as e-mail and chat sessions to an
appropriately skilled agent. Contact Center Express utilizes the powerful routing
algorithms resident in Avaya Communication Manager to determine the right
resource for the right interaction. Avaya Contact Center Express provides a set of
multi-channel capabilities that medium-sized contact centers can leverage and build
upon:
Desktop applications, including Agent Applications, Supervisor Applications,
and Utility Applications. These out-of-the-box applications allow you to begin
working with new technologies within hours.
Framework applications for the contact center, including Intelligent Routing,
Interaction Data, and Centralized Configuration.
December 1, 2006
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Page 274
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Part 1 – Section 8 - Contact Center
Multi-channel routing for voice, e-mail, and Web chat allowing you to create
true universal agents.
Outbound dialing with automated and agent-initiated Preview Contact.
Simple but effective, designed to solve costly outbound dialing issues, from
callbacks to targeted campaigns.
Powerful application development tools for complete customization and
integration capabilities.
Simple and fast wizards for desktop screen pops and routing rules.
Contact Center Express provides functionality that can easily and quickly adapt to
business dynamics without requiring a large budget and IT staff. Contact Center
Express is able to fully leverage the unique abilities of Avaya Communication
Manager, and provides multi-channel and agent performance enhancement
capabilities that translate into real results for your contact center.
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?
Avaya Response:
Comply. You can pre-record your desired waiting intervals on your internal
announcement hardware using the standard Recorded Announcements feature and
play the appropriate recording based on Avaya Communication Manager’s precise
prediction of Expected Wait Time (EWT). The announcement of Expected Wait
Time has NO impact on a caller’s state in queue. For example, you might
record the following announcements and select the recording with a goto step . . . if
expected wait . . .
Your estimated wait time is less than 1 minute
Your estimated wait time is 1-2 minutes
Your estimated wait time is 2-3 minutes
Your estimated wait time is 3-4 minutes
Your estimated wait time is 4-5 minutes
Your estimated wait time is greater than 5 minutes
Since callers usually expect an estimate expressed in minutes only, the internal
Recorded Announcement solution used in conjunction with Call Vectoring and the
Expected Wait Time algorithm is a common, economical solution.
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Avaya offers precise Expected Wait Time (EWT) announcement capabilities through
an optional Avaya Interactive Response (IR) system. Used in conjunction with Call
Vectoring and Dynamic Announcements, the IVR application receives the EWT from
Avaya Media Servers and can play back a dynamic announcement to the caller in
various formats; for example, “Your estimated wait time will be approximately 3
minutes 45 seconds.” The EWT Announcement is created dynamically and does not
have to be recorded multiple times as is the case when utilizing the internal
announcement hardware to provide expected wait time announcements.
How the Application Calculates Wait Time
The patented complex Expected Wait Time (EWT) algorithm calculates how long a
call has been or will be in queue. Previously, expected wait time was calculated
solely on historical data. The Avaya EWT algorithm analyzes the following factors on
a call-by-call basis to provide precise routing: call removal rate from the queue,
number of agents available, and queue length. It also considers priority queuing,
calls queued to multiple splits/skills, call abandons, time in Auxiliary Work, pending
agent moves, Direct Agent Calls, and agents in multiple splits/skills. Avaya’s
patented EWT algorithm encapsulates all of the dynamic factors which determine
the customer’s “wait” time experience for use by applications to deliver exceptional
customer service, single and MultiSite load balancing, and network cost savings.
Calculations can now be based on the expected wait time prior to queuing to a
skill/skill. Calls already in queue can be differentiated from new calls. Internal
announcements can give callers a range of estimated wait time in queue. This
information can also be passed to Avaya Interactive Response (IR) systems using
the Converse vector command to announce the precise expected wait time for
callers.
Our EWT algorithm is demonstrably more accurate than other predictors, and
actually predicts changes in wait times before they occur. EWT is responsive to
changing contact center conditions. For example, EWT adjusts instantly to any
staffing changes in the split, or if agents moves in or out of auxiliary work mode,
the wait time predictions immediately adjust. This predictive ability of EWT allows
call center managers to intervene and redirect calls to alternate treatments before
actual wait times exceed pre-established thresholds. Using EWT to redirect calls can
increase customer satisfaction, decrease costs, and create a more manageable call
center environment. Historical predictors tell you that you just had a problem; realtime predictors tell you that you are now having a problem; and EWT tells you that
you are about to have a problem. Only EWT allows you time to prevent the problem
from occurring.
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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.
Avaya Response:
Comply. You can use either Avaya Integrated Call Prompting or an Avaya
Interactive Response (IR) system to announce an expected wait time and
periodically offer callers the opportunity to transfer to a voice messaging
application. The call is not removed from queue in order to provide this
functionality. If the caller chooses to leave a message, normally the call is then
removed from queue and transferred to the messaging application to let them leave
a message for a callback and the caller will choose to hang up at that point.
If the caller wants to leave a message and continue to wait, then if the message is
left in an IVR callback messaging application, the application must make provisions
for this. Since the Avaya Call Center and the Avaya IR can communicate with the
Converse vector command, this allows a caller to be connected to the Avaya IR
while retaining its place in queue for the primary split/skill. This feature allows voice
response applications for the Avaya IR to make valuable use of caller wait time.
One of the strongest features of this voice response integration with the call center
is the ability to deliver self-service options to callers while waiting in queue for a
live agent. By providing the caller with useful options, the caller is better served,
and the call center manager can now manage peak queue volumes without hiring
additional expensive resources. Offer your callers a variety of customer self-service
options that make their calls more productive. IVR applications include information
bulletin boards, audiotex, form filling, transaction processing, dynamic
announcements, expected wait time announcements, custom call routing, and
callback messaging as examples.
If the caller wants to leave a message and continue to wait, then if the message is
left in an Avaya messaging solution such as Avaya Modular Messaging, then the
caller will have to elect to return to queue and the messaging system must transfer
them back to the queue. A special VDN can be setup to handle the transfer back if
it is desired that the caller re-enter the queue at a higher priority than before.
Normally, it is not preferred to allow the caller to both remain in queue and leave a
message since if the call is ultimately answered by an agent, then the message is
redundant. If the call is not answered by an agent before the caller abandons, then
the time spent waiting for the agent during and after leaving the message ties up
resources unnecessarily.
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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
Describe the system’s GUI administration tools.
Avaya Response:
Comply. Avaya Call Management System (CMS) Supervisor provides a GUI
administration interface for reconfiguring call control by editing vectors using Visual
Vectors graphical editor (described below) and changing agent assignments. With
CMS Supervisor, you can change agent split/skill assignments using the Change
Agent Skills window or the Multi-Agent Skill Change window. The Multi-Agent Skill
Change window facilitates quickly highlighting multiple agents (up to 32 at a time)
currently assigned in one skill group and dragging and dropping them to a new skill
group.
Supervisors can use the Variables in Vectors feature to program priority changes for
multiple calls simultaneously. Supervisors can view details about the number of
abandoned calls using CMS Supervisors real time monitoring and historical
reporting capabilities. Optional add ons such as Nice Analyst or Avaya Operational
Analyst can provide a cradle-to-grave tracking details for all calls including
abandons.
What-If modeling is provided by our Integrated Forecasting module, standard on
Avaya CMS, described in Response 8.4.0.
Avaya CMS Supervisor
The Avaya Call Management System (CMS) is accessible from direct, dial up, or
LAN-connected, Windows-based PCs using Avaya Call Management System
Supervisor software instead of (or in addition to) direct connected or dial up
dedicated terminals. CMS Supervisor eliminates the need for a dedicated terminal,
recovering desk space and reducing hardware investment, while delivering all the
advantages of a Windows and LAN environment.
The CMS Supervisor interface provides the following features and benefits:
Windows Graphical User Interface allows you to monitor and move multiple
agents easily with the use of a mouse versus a series of commands; it also
has the familiar look, feel, and increased efficiency of traditional Windows
features—point-and-click, drag-and-drop, and drop-down menus
Ability to run other PC applications while actively monitoring call center
conditions. You can run a report minimized and be notified (through
color/symbol changes) when an item has passed a specific threshold.
Enhanced, full color, graphical status reports can be generated in formats
that are easier to interpret at a glance.
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Customized threshold and exception alerting help call center managers
rapidly respond to changes within the call center
Utilizing existing PC and LAN environments results in cost savings, recovery
of desk space, and protects infrastructure investments by eliminating the
need for a separate terminal. Allows users to print reports on any network
printer for which the user has permissions.
Expanded mobility with access to CMS from the desktop or laptop PC, within
the call center, or from remote locations via dial-up access or local or wide
area network.
Access and monitor multiple call centers simultaneously. CMS Supervisor
supports multiple windows as well as multiple instances allowing CMS
Supervisor to connect to up to four different CMS platforms simultaneously. A
single CMS can support up to eight Avaya Call Centers.
Automatic execution of CMS reports and ACD administration and other tasks
with the Scripting feature. The Scripting feature provides another method to
automatically schedule and print reports, make ACD administration changes,
and perform other scheduled tasks. Scripting allows scheduling of call center
tasks such as agent reconfiguration, report generation, and vector routing
changes, with your PC scheduling package.
Fast, easy creation of customized reports with the new Avaya Report Wizard,
available with the optional Report Designer package, provides a wizard
approach for easy, customized report creation.
Easy export of call center data to other Windows applications via clipboard
cut and paste, exporting to a file, optional Open Database Connectivity, or
exporting to HTML (Hyper Text Markup Language) for posting your results on
your Intranet.
Avaya Visual Vectors
Avaya Visual Vectors, included with the Avaya Call Management System (CMS),
provides a graphical routing administration interface. With Avaya Visual Vectors,
you’ll be able to build even the most complex call-handling paths very quickly and
efficiently. Instead of working with traditional vector routing tables, you’ll be
creating a graphical “map”—a visual representation of your call distribution, with
familiar icons and graphics.
The Visual Vector Editor screen provides a palette of available functions and steps
that are grouped in logical sets. To build your call vector, you simply drag the
desired step or function into the appropriate place on the vector display grid. To
help you build the vector correctly and logically, the software automatically prompts
you to complete the logic of each step. If you add a decision or test-type step, for
example, the software will automatically create two or more branch paths for you—
which must be completed for the vector to be valid.
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Comments can be included with vector steps. For example, the text of an
announcement can be input as a comment for the announcement step. Comments
can be displayed for all steps, or can be viewed for a specific step by pausing the
cursor over the step icon. Free-floating comments can also be ‘pasted’ onto the
vector display grid, which can be useful when explaining the vectors to others, or
simply as development notes.
Avaya Visual Vector Editor
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.
Avaya Response:
Comply. Avaya offers IP Agent (described below) as a softphone application
designed specifically for Avaya call center agents. In addition, both our Contact
Center Express and Interaction Center multimedia contact center offers provide an
integrated agent softphone.
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Avaya IP Agent
Avaya IP Agent is a soft phone application that enables agents to work from any
PC, anywhere, as long as they can connect to your corporate network. Avaya IP
Agent provides the complete set of sophisticated agent features that you’ve come
to expect from Avaya’s best-in-class suite of contact center products, plus an
additional set of powerful capabilities.
Avaya IP Agent provides the first step towards SIP and Presence in the call center.
Agent productivity improvements are driven IP Agent’s ability to integrate and
interoperate with other applications and devices. It incorporates enterprise Instant
Messaging, provides a screen pop of customer-contextual data, interoperates with
Internet Explorer and Outlook to provide click-to-dial, and enhances DCP and IP
telephones, (including Callmaster IV and V terminals) by its ability to operate in
shared control mode.
IP Agent can address four separate client scenarios or requirements:
1. Allows a customer to extend their call center with a converged IP solution
and take the first step toward SIP and Presence.
2. Provides a PC-based agent softphone that allows agents to focus all their
attention on the PC, rather than splitting their time between two devices to
complete a transaction.
3. Delivers a sophisticated, full-featured remote agent solution.
4. Provides same user interface for remote and on-premise agents to help
reduce training time.
Avaya IP Agent integrates a flexible IP softphone client with a SIP/SIMPLE-based
Instant Messaging (IM) client. It incorporates a contact list of other IP Agent and IP
Softphone users and makes both phone and IM presence visible to other users. It is
simple to toggle between the softphone and IM applications. The IM and presence
capabilities require registering with the Avaya Converged Communications Server,
which is available separately.
IP Agent Shared Control licenses enable users who already have conventional
phones or Callmaster IV and V terminals to use the advanced communication
application features of IP Agent, such as instant messaging, screen pop, and
VuStats Monitor.
Avaya IP Agent accommodates all the Avaya call center agent features and
capabilities for agents working remotely or in an office location. Agents have access
to the full range of Avaya agent capabilities using an intuitive, customizable,
graphical user interface (GUI) using standard Microsoft Windows conventions.
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The highly customizable graphical user interface gives agents easy and immediate
access to customer care functions supported by Avaya Communication Manager on
the Avaya Media Server. Agents can customize their desktop by selecting their most
frequently used phone features for display in a separate window. In addition the
solution enables features such as integrated phone directories, last numbered
dialed, and screen pops with relevant customer data to enable more personalized
service.
The IP Agent solution includes an intuitive interface to access existing corporate
database information via LDAP (Lightweight Directory Access Protocol), and an
integrated contact history feature that allows agents a detailed view of the calls and
IMs made and received. In addition, contact center managers can administer screen
pops based on commonly used triggers, such as dialed number identification service
(DNIS), automatic number identification (ANI) and prompted digits.
Avaya IP Agent supports agent greetings. The agent can record, play, stop or erase
greetings through an easy to use menu. Up to 15 different greetings can be
recorded, each with a length of approximately 30 seconds. An option allows for the
agent greetings to be stored on a network drive rather than on the local agent PC,
which is ideal for call centers who utilize “hot seating” arrangements.
Avaya IP Agent will support seven configuration options including a dual-line mode
for separate voice and data, combining over one VoIP path, and sharing control
with an existing phone.
1. Telecommuter (previously called the dual connect)– one network connection
for the PC and one telephone connection. Supports agent greetings via using
Avaya Switcher II
2. Road Warrior (VoIP) – one network connection for the PC to access the
Avaya communication server. Agent greetings are stored in the PC
3. Terminal Services – one network connection for the client device and one
telephone connection. No greetings. No IM
4. Avaya Telephone (DCP and IP) – one network connection and one DCP/IP
telephone connection (DCP: 2400 or 6400, IP: 4600 or 9600 Series
telephones), one network connection for the PC to access Avaya
communication system. No agent greetings.
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5. IP Telephone – one TCP/IP network connection for the PC and one IP network
connection for the IP telephone. Doesn’t support greetings.
6. CallMaster VI – one DCP connection to the Avaya communication server and
a serial (RS-232) connection between the PC and the Callmaster VI
telephone. Greetings stored on the CallMaster VI. No IM.
7. Instant Messaging – Avaya IP Agent integrates a flexible IP softphone client
with a SIP/SIMPLE-based Instant Messaging (IM) client. It incorporates a
contact list of other IP Agent and IP Softphone users and makes both phone
and IM presence visible to other users. It is simple to toggle between the
softphone and IM applications. The IM and presence capabilities require
registering with the Avaya Converged Communications Server, which is
available separately.
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Avaya is creating and executing one of the best suites of Contact Center solutions
in the industry. The IP Agent application will interact with and access these solution
features and capabilities and bring them wherever an enterprise chooses to put its
workforce - at home or in the office.
System Requirements
Operating System for IP Agent desktop: Microsoft® Windows® 2000
Professional for Intel x86 processors or Microsoft Windows XP Professional
PC Configuration: Intel® Pentium® III 300 MHz or higher PC, 30 MB of
available hard disk space, Minimum of 128 MB RAM, Full-duplex sound
card, headset, microphone, Microsoft Internet Explorer 5.5 SP2 or higher
Avaya PBX release: Avaya DEFINITY® 10, Avaya MultiVantage™ 1.1 or
1.2, Avaya Communication Manager 1.3 or higher
Call Center software release: Avaya Call Center R9 or later
Advanced Segmentation (AS) Screen Pop requires
Manager 3.0 or later with Advanced Segmentation.
Communication
Instant Messaging requires Converged Communication Server 2.1 or later
Shared Control of CallMaster IV and V terminals requires Communication
Manager 3.0 or later
Agent Greetings in telecommuter mode requires the Avaya Switcher II
adapter.
Agent Greetings
Avaya IP Agent will support seven configuration options. Agent greetings are
supported in the Telecommuter, Road Warrior, Shared Control, and Callmaster VI
modes as described below.
Telecommuter – one network connection for the PC and one telephone
connection. Supports agent greetings via using Avaya Switcher II.
Road Warrior (VoIP) – one network connection for the PC to access the
Avaya communication server. Agent greetings are stored in the PC.
Terminal Services – one network connection for the client device and one
telephone connection. No greetings. No IM
Shared Control - Avaya Telephone (DCP and IP) – one network connection
and one DCP/IP telephone connection (DCP: 2400 or 6400, IP: 4600 or 9600
Series telephones), one network connection for the PC to access Avaya
communication system Supports agent greetings via using Avaya Switcher
II.
IP Telephone – one TCP/IP network connection for the PC and one IP network
connection for the IP telephone. Doesn’t support greetings.
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CallMaster VI – one DCP connection to the Avaya communication server and
a serial (RS-232) connection between the PC and the CallMaster VI
telephone. Greetings stored on the CallMaster VI. No IM.
Avaya IP Agent supports agent greetings for the Road Warrior, Telecommuter,
Shared Control, and CallMaster VI configurations. (Telecommuter and Shared
Control use the Avaya Switcher II adapter.)
In the VoIP Telecommuter, and Shared Control configurations, the agent greetings
are stored as .wav files on the agent's PC. Up to 15 different greetings can be
recorded, each with a length of approximately 30 seconds. For the Telecommuter
configuration, an additional piece of hardware is required, the Avaya Switcher II.
In the CALLMASTER VI configuration, the agent greetings are stored on the internal
announcement unit within the voice terminal. Up to six different greetings can be
stored, each with an approximate length of 9 seconds.
Neither the IP Telephone configuration nor the Windows Terminal Services/Citrix
configuration supports agent greetings.
IP Agent will provide support for selecting the appropriate greeting when receiving
an incoming call. The user (e.g. agent or call center administrator) will be able to
administer which greeting is played based on the login status, agent state, agent id,
prompted digits, and ANI or VDN.
Starting with IP Agent R5, a new program option has been added to allow for the
agent greetings to be stored on a network drive rather than on the local agent PC.
This feature is ideal for call centers who utilize “hot seating” arrangements.
Agent greeting can be accessed on the main window with the Agent Greeting
toolbar. The Agent Greeting toolbar allows the agent to select a greeting, play, and
stop the greeting. The greetings are administered on the Agent Greetings window.
The window allows the user to record, play, stop, and erase or delete a greeting.
Agent Greetings Window
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8.1.7 ACD Voice Terminal
IP desktop voice terminal instruments will be required for agent positions.
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.
Avaya Response:
Comply. All Avaya voice terminals, digital, IP, or analog can be used for ACD
agents. The recommended voice terminal is dependent upon the feature
requirements for your agents and which technology you choose.
Avaya 9600/4600 Series IP Telephones
The Avaya 9600/4600 Series IP Telephones deliver an extensive set of software
features, high audio quality, and attractive streamlined design. Advanced webenabled graphical displays on the 9600 Series as well as the 4610SW, 4621SW,
4622SW and 4625SW support browser-based desktop applications such as online
order entry and inventory lookup in addition to more traditional voice applications
such as directory-based dialing and call logging. With the introduction
Communication Manager R3.1, the 9600/4600 Series IP Telephones become more
functional application platforms. Both telephone series support third party
applications that can push content to the displays or audio path through the
application programming interfaces (APIs) on these phones. For example,
emergency alerts and other applications can be supported.
Avaya 9600/4600 Series IP Telephones are simple to use with both fixed and
flexible feature buttons, easy-to-read graphics, and several wall mount and desk
mount options. They have been optimized for reliable use in IP networks, with
sophisticated security capabilities such as media encryption and protection from
denial of service attacks. Built-in Ethernet switch ports enable streamlined desktop
implementations, while voice packets are tagged with the appropriate quality of
service (QoS) parameters such as 802.1q and DiffServ for priority treatment by
QoS-enabled IP networks. The 9600/4600 Series IP Telephones also support the
802.3af power over Ethernet standard.
The 4610SW IP Telephone provides a medium screen graphic display, paperless
button labels, call log, speed dial, 12 programmable feature keys, Web browser,
and full duplex speakerphone. It also includes a two-port Ethernet switch. The
4610SW supports Unicode with R2.1or higher firmware.
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Avaya 4610SW IP Telephone
The Avaya 4621 and 4622 IP Telephone is cost effective and provides a large
screen graphic display, paperless button labels, call log, speed dial, 24
programmable feature keys, Web browser, and full duplex speakerphone. The
4621SW also includes a 2 port Ethernet switch.
Avaya 4620SW& 4621SW IP Telephone
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Avaya 4622SW IP Telephone
Avaya CallMaster® V Digital Voice Terminal
The Avaya CallMaster® V has been specially designed to support applications
involving the Automatic Call Distribution (ACD) feature of the Avaya Communication
Manager. The ergonomic design of the CallMaster V enables agents to handle large
volumes of calls more quickly, efficiently, and productively—in customer service,
order processing, collections, account management, or any communicationsintensive activity. VuStats’ display of agent and call center statistics on the
CallMaster V provides agents with real-time information they can use to improve
their own performance and that of the call center.
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The CallMaster V has the same look and feel of the standard Avaya 6400 Series
telephones. There are two significant additional features that maximize the value of
this telephone in a Call Center environment.
2-Built-in Headset Jacks – The CallMaster V is designed to use Avaya
headsets.
Built-in Recorder Interface Module (RIM) with Warning Tone – Will support
recording of both the agent’s and caller’s voice on a voice activated analog
tape recorder. A soft beep warning tone is repeated every 13.5 seconds to
notify the agent and calling party that the call is being recorded (user can
deactivate).
CallMaster V is designed to work on a 16- or 24-port, 2-wire Digital Line Circuit
Card.
The CallMaster V digital terminal is also equipped with the following:
16 Dual LED call appearance/feature buttons
10 Fixed features – Speaker, Mute, Conference, Transfer, Hold, Redial, Menu,
Exit, Previous, and Next
Adjustable 48 character (2-lines by 24 characters) Liquid Crystal Display
which provides agents with display of ACD messages, unified messaging
access, and call-related information, including Dialed Number Identification
Service (DNIS), Automatic Number Identification (ANI), and VuStats
12 Assignable soft key features associated with the display
Built-in one-way, listen-only speaker for group listening, on-hook dialing, or
hands-free listening to voice mail
Adjustable volume control (handset, speaker, and ringer)
Station users may be allowed to program, remove, or rearrange the following
features on set:
Account Code Entry
Directed Call Pickup
Automatic Dialing Buttons
Group Page
Blank (to remove feature)
Send All Calls
Call Forward
Whisper Page
Call Park
Whisper Page Answer
Call Pickup
Whisper Page Off
The System Administrator may substitute other soft key features for the above:
12-Button touch-tone dial pad with raised bar on “5” for the visually impaired
Message waiting light (LED)
Eight personalized ringing options
Seven foot modular line cord
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Pull-out feature reference card tray
Stand for desk or wall-mount configuration
International portability
Downloadable transmission parameters
Additional options include:
XM24 Expansion Module with 24 buttons, increasing the total button capacity
to 40 buttons. All 24 buttons have dual LED lights and can be administered
for either call appearances or features.
K-Type handset with nine foot modular cord
12-Foot modular handset cord
14-Foot and 25-foot modular line cords
Avaya/Plantronics headsets
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.
Avaya Response:
Comply. Avaya Call Management System (CMS) Supervisor software for your
supervisor PCs provides access to all standard and custom reports, graphical and
text-based, available under CMS. And now, these reports are even better with the
following customization enhancements:
Sorting capabilities that let supervisors rearrange report information in an
order that is easy for them to use. Each supervisor can customize reports
with up to three sorting levels, such as alphabetically, by agent/state,
numerically by time/agent, and more.
More options for call center analysis with the capability to mix real-time and
historical information together on the same standard report.
New Graphical Supervisor reports are easier to interpret and more
meaningful thanks to improved graphical display capabilities, including
colors.
Ability to customize reports by selecting data, formatting, positioning, and
the type of charts (including pie charts, 3-D bar charts, line graphs, and
simple grids).
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Customized threshold and exception alerting help call center managers
rapidly respond to changes within the call center
CMS Supervisor lets you create call center reports in the colors and format
that work best for you, and import reports or data between CMS and other
applications as needed
Avaya Call Management System (CMS) Supervisor real-time reports give
supervisors snapshots of the call center’s performance and status. Abandoned calls,
for example, can be monitored to determine the waiting-for-service tolerance of
callers and compared to the number of calls in queue. Additionally, agent
productivity can be compared at a glance to determine who may need help in
speeding after-call work.
Over 40 real-time reports are available in a variety of easy-to-interpret graphical
and text-based formats that can be displayed on your PC, printed, stored to a file,
copied to a clipboard, run as a script, or exported to HTML format through the Save
as HTML feature. Standard real-time reports display data for the current interval for
agent, split/skill, trunk/trunk group, vector, and VDN activities, such as number of
ACD calls, abandoned calls, average talk time, and so on.
You can use Avaya CMS’ reporting capability to get real-time information—updated
as often as every three seconds, depending on the user permission and number of
active terminals and open windows—that will help you monitor ongoing
performance and status so you can make any necessary adjustments quickly.
Avaya CMS Split/Skill Status Real Time Report
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In the CMS Supervisor real time skill status display shown below the chart format
and sorting order of agents has been changed from the pie chart shown above and
the agents are sorted in ascending time order. Threshold highlighting is also
illustrated showing agents spending excess time in Aux Work state, a Service Level
Warning, and an Oldest Call Waiting caution level.
Avaya CMS Split/Skill Status Real Time Report
Double clicking on
an agent name in
the report allows a
supervisor to drill
down and access
individual agent
information.
Avaya CMS Agent Information Drill Down Report
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Avaya Operational Analyst
Avaya Operational Analyst is a full-featured multi-channel contact center real-time
performance monitoring, historical reporting and sophisticated analytical system
supporting Enterprise businesses that need integrated and consolidated operational
data storage, reporting and analysis for the Contact Center. Avaya Operational
Analyst is an optional component of Interaction Center, functioning as its
operational data store and contact center performance analysis system, provide
reporting and real time monitoring.
The integrated customer interaction repository and suite of standard predefined
reports provide detail and summary views across multiple sites, multiple ACD
vendors and multiple communication channels (Voice, VoIP, email, Web Chat and
Web Self-Service). The 3-D graphical, web-based monitoring of real time and
historical contact center performance allows for efficient management of agent
activities and verification that the system is achieving service levels goals. The
ability to create and maintain customized reports aids in the assessment of
customer service and marketing activities.
The Operational Analyst Real-time Event Processor collects and processes real time
events from an IC system. Data from agent desktops and the multi-media channels
are collected and processed to provide comprehensive and consistent real-time
reports across all channels. In addition, different views of real-time data are
provided including the current 30-minute interval and up to 4, user-defined 24-hour
views.
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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.
Avaya Response:
Comply. Agents can receive real time display information using VuStats on the hard
phone or Softphone as described below. Avaya Contact Center Express also
provides displays on the agent’s desktop for display of real time information as
described below. Avaya Interaction Center can be customized to provide ACD
statistics on the desktop.
VuStats on the Agent Telephone Display
Avaya offers the VuStats feature for display of ACD statistics on the agent voice
terminal. VuStats is a convenient, cost effective way for call centers to measure
results in real time. Anyone with a display-equipped voice terminal, including call
center managers and non-ACD personnel, can use VuStats to view real-time or
cumulative daily call center statistics. VuStats gives agents the power to judge their
own performance and take steps to modify call handling skills to improve
productivity.
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For example, agents with VuStats can:
View calls in queue and/or wait times to delay nonessential activities until call
delays are acceptable
View their time in Auxiliary Work
Compare their productivity with call center objectives or the performance of
other agents and know when to step up the pace
Keep track of their total cumulative performance for an entire day
Be automatically notified by a flashing lamp when thresholds are reached for
individuals and groups
Up to 50 different 40-character display formats (each with up to 10 fields of data)
can be customized, thereby creating displays of information that are important to
call center personnel. Thresholds can be defined on data items that will cause the
VuStats lamp to flash when the displayed item exceeds a pre-defined threshold. All
data is cumulative up to the current second, combining current interval and
historical data. Most data can be cumulative for the entire day or for the most
recent 24 hours or half hours. Redisplay formats can be linked so the agents can
step through a series of displays to view their progress against different
measurements.
The VuStats feature supports display of the following ACD statistics on the agent
voice terminal:
VuStats Data Items
ACD calls
agent extension
agent name
agent state
average ACD call time
average ACD talk time
average extension time
call rate
current reason code
current reason code name
elapsed time in state
extension calls
extension incoming calls
extension outgoing calls
shift ACD calls
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Agent and Agent Extension Data Types
skill level
split calls flowed out
split acceptable service level
split calls waiting
split ACD calls
split extension
split after call sessions
split name
split agents in other
split number
split agents on ACD calls
split objective
split agents on extension calls
split oldest
call waiting
split agents staffed
split percent in service level
split average ACD
split total ACD
talk time
talk time
split agents available
split total
after call time
split agents in after call
split total aux time
split agents in aux (1-9, all, default, nontotal ACD call time
default)
split average
total ACD talk time
after call time
split average
total after call time
speed of answer
split average
total aux time
time to abandon
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VuStats Data Items
shift aux time (1-9, all, default,
non-default)
shift aux time
reason code
shift average ACD
talk time
Split/Skill Data Types
acceptable service level
ACD calls
after call sessions
agents available
agents in after call
agents in aux (1-9,all,default,nondefault)
agents in other
agents on ACD calls
agents on extension calls
VDN Data Types
acceptable service level
ACD calls
average ACD talk time
average speed of answer
average time to abandon
calls abandoned
average incoming
call time
average outgoing call time
incoming abandoned calls
incoming calls
incoming usage
Agent and Agent Extension Data Types
split call rate
total available time
split calls abandoned
total hold time
split calls flowed in
total staffed time
agents staffed
average ACD talk time
average after call time
average speed of answer
average time to abandon
call rate
calls waiting
oldest call waiting
percent in service level split
extension
split name
split number
split objective
calls abandoned
calls flowed in
calls flowed out
total ACD talk time
total after call time
total aux time
calls flowed out
calls forced busy
or disconnected
calls offered
calls waiting
non ACD connected calls
oldest call waiting
percent in
service level
total ACD talk time
VDN extension
VDN name
Trunk Group Data Types
number of trunks
outgoing calls
outgoing completed calls
outgoing usage
percent all trunks busy
percent trunks
maintenance busy
trunk group name
trunk group number
trunks in use
trunks maintenance busy
VuStats Support on Avaya IP Agent Softphone
Agents and administrators can check the VuStats Monitor window for information
on Contact Center operations, like the number of calls waiting for a particular split.
The information is updated periodically, based on pre-defined refresh rates that
users can select. Agents can automatically monitor one or more lines of VuStats
information at the same time. The information will be presented in a window
independent of the main window.
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The user can administer the refresh rate for the VuStats screen. All the VuStats
displays can be updated every 10, 20, 30, 60 and 120 seconds. The user can also
administer the display interval per button between the display updates (i.e. the
time the monitor waits to gather display information before moves to the next
VuStats display). The display intervals can be 1, 3, 5 and 10 seconds. Intervals
may need to be adjusted based upon network speed.
VuStats Monitor
Contact Center Express PC Wallboards Keep Your Agents Informed
Wallboard is a Windows-based application that displays real-time statistical
information on VDNs, skills or splits and agents in a marquee window. Installed on
agent PCs, the scroll bar of information allows agents to closely track their personal
work performance and the performance of their work group (skill or split).
Statistical information is sent to the Wallboard application from the Interaction Data
Server, which monitors VDNs, splits, skills and agent extensions, and then
calculates statistics about all facets of a call.
This powerful feature is easy to administer. It eliminates the need for a separate
expensive wallboard within a call center. Custom messages can be added with color
and flash. For example, if an inbound call center wants to feature their TVs at a
special customer price, that information could be flashed on the wallboard in red.
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Wallboard can be used as an Agent extension or a standalone application. The
following types of information can be displayed on the wallboard:
Wallboard VDN Information
o
VDN names and numbers
o
The number of calls waiting to be answered for the specified VDN
o
The length of time the first call in the queue has been waiting (in secs)
o
The average length of time agents are talking to callers to this VDN (in
secs)
o
The average length of time callers to this VDN are waiting before their
call is answered (in secs)
o
The number of calls to this VDN that have been abandoned
o
The average length of time callers to this VDN are waiting before
abandoning their calls (in secs)
Wallboard Skill or Split Information
o
Skill or split names and extension numbers
o
The number of calls waiting to be answered for the specified skill or
split
o
The number of agents logged into the skill or split that are available to
take calls
o
The number of agents logged into the skill or split that are unavailable
to take calls
o
The average length of time agents logged into the skill or split are
talking to callers (in secs)
o
The average length of time callers to this skill or split are waiting
before their call is answered (in secs)
o
The number of calls made to the skill or split
o
The number of calls to the skill or split that have been abandoned
o
The average length of time callers to this skill or split are waiting
before abandoning their calls (in secs)
Wallboard Agent Information
o
Agent names, IDs and station numbers
o
The current work mode of the specified agent
o
The number of calls the agent takes per hour
o
The average length of time the agent spends on a call (in secs)
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o
The average length of time the agent spends in After Call Work mode
(in secs)
o
The average length of time the agent spends in Auxiliary mode (in
secs)
o
The agent's pending work mode
o
The agent's pending reason code
o
The last reason code the agent used
o
The skill or split group the agent is logged into
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.
Avaya Response:
Comply.
Avaya Call Management System (CMS)
The Avaya Call Management System (CMS) includes:
43 Real Time standard reports covering: Agent, Events, Multi-ACD, Queue,
Split/Skill, Trunk Group, VDN, Vector, and Drill-down reports.
7 Integrated standard reports covering: Agent, Split/Skill, and VDN.
100+ Historical standard reports covering: Agent, Agent Attendance,
Login/Logout, Reason Codes, Inbound/Outbound, Agent Trace, MultiACD,
Events, Call Records, Call Work Codes, Split/Skill, System, Trunk,/Trunk
Group, VDN, Vector, Busy Hour, Forecasting, and more.
Custom Reporting options are also available.
Reports are available: real-time, integrated, historical, and custom, on
demand or scheduled, text-based and graphical, available on the PC display,
can be printed, saved to file or exported to HTML formats for posting on a
web server.
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Avaya Call Management System (CMS) statistics for agents, split/skills, trunks,
trunk groups, vectors, and VDNs are stored in customer-defined intervals (15, 30,
or 60 minutes system-wide) for up to 62 days. Daily statistics are stored in 24-hour
intervals for up to five years; weekly and monthly summary data can be stored for
up to ten years. Up to 2,000 customer-defined exceptions are saved; and up to 15
days of Special Days forecasting information can be stored. The raw historical
interval data is stored and used for the ad hoc generation of historical reports
covering any period of time within the storage intervals. The system supports
expandable storage capacities to provide long-term data storage up to these
system maximums.
The Avaya Call Management System (CMS) database engine is an ODBC complaint
Informix Dynamic Server (IDS) formerly called On-Line engine. R13 CMS will use
IDS version 9.4. IDS is fully supported by Informix, provides improved performance
and improved database corruption protection, and supports much greater file sizes
(>2 GB). Support for non-disruptive backup and restore is provided. CMS supports
an optional ODBC interface using a standard off-the-shelf OpenLink ODBC driver.
Avaya Operational Analyst
The multichannel Customer Interaction Repository features a common catalog of
detailed customer data that can contain multichannel data from Avaya Interaction
Center and voice data from Avaya Call Management System (CMS). The term
Customer Interaction Repository refers to a collection of database tables that is
used to record summarized information about activities in your contact center.
The Avaya Operational Analyst (OA) Basic and Advanced Reports as well as
additional reporting tools provided by Avaya IC can be used to report on data in the
shared repository.
The following customer provided databases are supported on OA 7.1:
Windows:
Microsoft SQL Server 2000 SP3a (Standard or Enterprise Edition, 32-bit
version)
Oracle 9.2.0 Patch 4.0 (Standard or Enterprise Edition, 32-bit version)
Oracle 10g
Solaris
Oracle 9i (32 bit)
Oracle 10g (64 bit)
AIX
IBM DB2 8.1, FP5 Enterprise Edition (32-bit version)
IBM DB2 8.2 Enterprise Edition (32-bit or 64-bit version)
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Basic Reporting Package
The Basic Report Package is designed for contact center supervisors with
performance and task-level priorities. A browser-based interface provides reporting
across all channels, with data presented in clear, compelling three-dimensional
graphics for rapid recognition of details. The Basic Reports also present data in
tabular, sortable format and utilize the technology of a third party tool from Visual
Insights to produce both real time monitoring and Avaya value added predefined
historical reports. This tool is useful for daily contact center operational reporting
and for providing real time monitoring capabilities. The Basic Report Package
satisfies the particular needs of the mid market customer for a single reporting tool.
Available reports include predefined historical reports across Avaya Interaction
Center, Avaya CMS—specifically External Call History and summary interval data—
and real-time monitoring and historical performance analysis for both agents and
skills. With the ability to refine reports down to contact detail, supervisors can
perform true cradle-to-grave analysis.
Real Time Monitoring and Predefined Reports
Real time monitoring permits the contact center supervisor to track real-time agent
and contact activity across Interaction Center channels to determine the
bottlenecks and quickly adjust agent schedules accordingly.
Real Time Service Class and Queue Status Report – This report keeps the
user informed of the real-time performance data for specified Service Class
or queue. The report also shows performance trending over a 30-minutes
interval.
Statistics for each login agent are collected for: Number of Work Items in
Queue, Expected Wait Time, Oldest Wait Time, Average Wait Time, 30Minute Average Wait Time.
Real Time Service Class and Queue Performance Report – This report
provides a more detailed view of how Service Class and Queues are
performing on the basis of a user-selected statistics.
Statistics displayed for this report are: Percentage of Work Items Handled
Within Service Level, Number of Work Items in Queue, Offered, and
Completed, Average Wait Time and Number of Abandoned Work Items.
Real Time Agent Time in State Report – This tabular report lets the
supervisor determine what state Agents are in, how long they’ve been in that
state, what their role is relative to a Service Class, and whether they can be
used to support other Service Classes or Queues.
Statistics displayed for this report are: Agent, State, Time in State, Service
Class or Queue, and Role.
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Real Time Agent Performance Report – This report provides a view of
individual agent work performance in real-time compared to absolute targets
or compared with other agents.
Statistics displayed for this report are: Agent, Number of work items
currently opened, Average Time Spent Working on Items, Average Time
Spent Wrapping up Work, Time Agent Spent in Idle State while Available,
Time Agent Spent on Break
Real Time Agent Performance by Job Report – This report provides an
operational view of multiple agents across multiple jobs. This allows the
supervisor to compare an agent’s performance to other agents working the
same job, or a single agent’s performance over several job categories.
Statistics displayed for this report are: Average Work Duration, Average
Wrap-Up Duration, Total Work Duration, Total Wrap-Up Duration, Number of
Work Items Completed, Number of Work Items Rejected, Number of Work
Items Previewed
Real Time Agent Set Outcome Codes Report (Outbound) – This report
provides an operational view of the distribution of outcome codes applied to
specific jobs by specific agents for the current interval. Any outcome codes
not selected in the report input page are summed and reported together.
Statistics from this report can help the supervisor determine which agents
are more effective in achieving successful outcomes.
Real Time Job Performance Report – This report provides an operational view
of how jobs are performing on the basis of a selected statistic. Performance
can be compared to absolute targets or other jobs.
Statistics displayed for this report is: Hit Rate, Average Work Duration,
Average Wrap-Up Duration and Nuisance Call Rate.
Real Time Telephone Number States Report (Outbound) – This report
provides a real-time view of how many numbers are in a particular state for a
set of jobs. Any states not selected in the report input are summed and
reported together.
Statistics from this report can help the supervisor determine how many
telephone numbers are considered unreachable for the job and how many
telephone numbers are scheduled for a call back?
Real Time System Set Completion Codes Report (Outbound) – This report
provides an operational view of the results of call attempts in the current
interval. Any completion codes not selected in the report input page are
summed and reported together.
Statistics from this report can help the supervisor determine how many calls
are reaching answering machines and how many nuisance calls have
occurred for each job?
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Real Time Agent Performance by Service Class and Queue Report – This
report provides an operational view of multiple agents work performance
across multiple service classes and queues. An agent’s performance can be
compared against other agents or when working in other service classes and
queues.
Statistics displayed for this report are: Agent, Average Work Duration,
Average Wrap-up Duration, Number of Work Items Opened, Number of Work
Items Completed, Average Customer Hold Duration and Average Deferred
Duration.
Predefined Historical Reports
Historical reports provide a way to assess call center efficiency and allow for
identification of trends and patterns in the data.
Historical Service Class and Queue Volume Report – This report provides a
way to assess where the bottlenecks are in the call center processes.
o
This report plots new work items compared to departed work items by
Service Class and Queue, shows average wait time by Service Class
and Queue.
Historical Service Class and Queue Performance Report – This report provides
a more detailed view of how Service Class and Queues are performing on the
basis of a user-selected statistics.
o
Statistics displayed for this report are: Percentage of Work Items
Handled Within Service Level, Number of Work Items Offered, and
Completed, Average Wait Time, Number of Abandoned Work Items,
and Average Time to Abandon.
Historical IC Agent Performance Report – This report provides a view of
individual agent work performance compared to absolute targets or
compared with other agents.
o
Statistics displayed for this report are: Agent, Number of Work Items
Opened, Number of Work Items Completed, Average Work Duration,
Average Wrap-Up Duration
Historical Agent Performance by Service Class and Queue Report – This
report provides an operational view of multiple agents work performance
across multiple service classes and queues. An agent’s performance can be
compared against other agents or when working in other service classes and
queues.
o
Statistics displayed for this report are: Agent, Average Work Duration,
Average Wrap-up Duration, Number of Work Items Opened, Number of
Work Items Completed, Average Customer Hold Duration and Average
Deferred Duration.
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Work Item Detail Report (combined IC and CMS data) – This report provides
detail contact history information. From this report, you can drill down to
Segment Information and Wrap Up Codes reports. This reports helps answer
the question of what was the customer experience when contacting the
center, how many times on average was a work item of a particular type
handled, deferred or put on hold.
CMS Detail Report - This report provides detail contact history information
based on CMS call history detail records. From this report, you can drill down
to Segment Information and Call Work Codes reports. The reports act as a
hierarchy with Call Detail reports displaying multiple calls, each with a
corresponding Segment Information report. The Segment Information
reports list multiple call segments for a particular call – with each segment
having a corresponding Call Work Codes report.
Historical Agent Performance (CMS) Report – This report provides an
historical view of CMS skill, allowing trend-based comparison of call counts
and average durations.
o
Statistics displayed for this report are: Number of ACD Calls, Average
ACD Duration, and Average ACW Duration.
Historical Agent Performance by Skill (CMS) Report – This report provides an
historical view of multiple agents and a single CMS skill, or a single agent and
multiple CMS skills, over time. An agent’s performance can be compared to
other agents.
o
Statistics displayed for this report are: Number of ACD Calls, Average
ACD Duration, and Average ACW Duration.
Historical Job Performance Report (Outbound) – This report provides an
historical view of outbound job performance over a period of time.
o
Statistics displayed for this report is: Hit Rate, Average Work Duration,
Average Wrap-Up Duration and Nuisance Call Rate.
Historical Agent Performance by Job Report (Oubound) – This report
provides an historical view of multiple agents and a single job or multiple
jobs and a single agent over a period of time.
o
Statistics displayed for this report are: Average Work Duration,
Average Wrap-Up Duration, Total Work Duration, Total Wrap-Up
duration, Average Preview Duration, Number of Work Items Completed
or Rejected.
Historical Agent Set Outcome Codes Report (Outbound) – This report
provides an historical view of the outcomes assigned by agents to outbound
calls over a period of time.
o
Statistics from this report can help the supervisor determine which
agents are more effective in achieving successful outcomes.
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Historical System Set Completion Codes Report (Outbound) – This report
provides an historical view of what happens to call attempts over a period of
time.
o
Statistics from this report can help the supervisor determine how
many calls are reaching answering machines and how many nuisance
calls have occurred for each job?
Historical Skill Performance Report – This report allows the supervisor to
compare statistics among Skills over a period of time.
o
Statistics from this report can help the supervisor determine which
agents are more effective in achieving successful outcomes.
Basic Report Customization
All colors (with the possible exception of the background gradient and the floor/wall
borders) are changeable by the administrator by editing the resource file for the
report. All aspects of text strings (font family, point size, content) can be changed
using the same mechanism since this is required for localization. Care must be
taken doing this, since these are in localized UTF8 format.
Additional Basic Reports (real time and historical) may be created using the Visual
Insights In3D Java-based developer’s toolkit. This toolkit must be purchased from
Visual Insights. For slight modification to one of the pre-built Basic Report, the
appropriate report template may be copied and quickly modified. For brand new
reports, one would use the Java toolkit and develop the report from scratch. The
data model is fully documented to aid in this effort. Basic Report modification and
creation is only recommended for programmers, i.e. Professional Services (BCSI),
Business Partners and System Integrators, not the call center supervisors.
Advanced Reporting Package
The Advanced Reporting Package is designed for sophisticated users and business
analysts who need to track key historical performance indicators and trends for
operational improvement. A windows or browser-based interface provides data in
analytical “cubes”— multi-dimensional graphic representations of data.
Cubes may be manipulated with a straightforward graphical tool to produce various
perspectives on the data, and on-screen performance metrics illustrate the business
value of each interaction.
The Advanced Reporting Package has two components, the reports and the
reporting tools. The Advanced Reports are the Avaya added value predefined
business value OLAP (online analytical processing) reports and provide historical
analysis on each Interaction Center channel as well as reporting for the IC Business
Applications. The Advanced reporting tools are based on Cognos technology and are
used for ad hoc querying, modifying reports, custom report creation and the ability
to insert custom calculations. With the ability to click on graphic elements and drill
down to supporting transaction detail, the user can perform all levels of
sophisticated business analysis.
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The Advanced Reporting Package is available on Windows or Web-based. The
Cognos tools include Powerplay and Impromptu. Powerplay is used for extracting
data into the multidimensional cubes and viewing the data with the predefined
reports. Impromptu is typically for advanced users who require direct access to the
data or who wish to create custom reports. Both windows and web-based versions
of the Cognos tools are available and orderable.
Multidimensional Cubes
Multidimensional cube, also known as a Multi-dimensional On Line Analytical
Processing (MOLAP) contains data on multichannel contact center statistics. It is a
summary of cumulative information about activities and transactions performed by
agents over specified periods of time. The cube allows for call center managers to
look at trends in the call center, such as the most frequently used media for
contacts. Information in the cube can be viewed in different combinations of
measures and dimensions, and in a variety of formats, such as tabular, line graph,
bar graph, pie chart or multi-dimensional graph.
There are 4 cubes created for the Advanced Reporting package:
1. Contacts Cube – contains data on contacts arrival and identification and
contact center’s response to contacts
Dimensions: Dates, Days of the Week, Time Ranges, Media types,
Number of Agents per Contact, Agents, Dispositions, Queues
Measures: number of contacts offered, number of contacts handled,
number of abandoned contacts, number of agent interactions, average
contact duration, average queue time, average answer time, average talk
time, average wrap-up time, average defer time
2. CMS cube – contains performance data collected from External Call History and
detailed data from CMS
Dimensions: Start Time, Time of Day, Day of Week, VDN, Skill
Measures: Wait Time, ACW Time, Talk Time, Number of Abandons,
Number of Answers, Number of Calls, Average Wait Time, Average Talk
Time, Average ACW Time, Percentage of Abandons, Percentage of Calls
3. Contact Segment (mma) cube
Dimensions: Date, Time of Day, Day of Week, Queues, Agent Names,
Service Classes, Media Type
Measures: Number of Contacts, Number of Abandons from Queues,
Percent of Abandons from Queue, Number of Abandons from Service
Class, Percent of Abandons from Service Class, Average Wait Time,
Average Work Time, Average Wrap-up Time, Percent Contacts
Redirected, Average Hold Time, Number Abandons from Hold, Percent
Abandons at Agent, Number Voice Contact Segment
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4. CallCenterQ (business apps) cube
Dimensions: Date, Time Range, Days of Week, Queues, Agents,
Owners, Workgroups, Contacts, Requests, Returns, Products,
Categories, Orders
Measures: Number of Contacts, Number of Requests, Number of
Fulfillment, Number of Orders, Number of Returns, Total Tasks
Handled, Fulfillment Order Total, Order Total, Quantity Ordered, Total
Owned Requests, Total Owned Fulfillments total Owned Orders, Total
Owned Returns, Total Tasks Owned, Total Assigned Requests.
Predefined Advanced Reports
A set of predefined views of the cubes (reports) is provided with Operational
Analyst. The reports can also be further drilled down to detailed ad-hoc reports.
Note: The number of PowerPlay reports is not limited to those listed here. The total
number of reports, or filtered views of the cubes, are limited only by the Number of
defined Dimensions * Number of Measures.
Predefined PowerPlay reports for Contacts cube
Abandoned Calls by Time of Day
Contacts Abandoned While Ringing or On Hold
Agent Interactions by Agent Group and Media type
Contacts handled by Automatic Agent
Agent interactions by Time of Day for Agent Groups and Phone media type
Contacts Offered by Time of Day for all media types
Number of Contacts offered by Time of Day and Day of Week
Average Queue Time per interaction for contacts
Drill-Through Impromptu Reports for Contacts cube
Drill_agent – provides detail info for the agent
Drill_Contact - provides detail info for the contact
Predefined PowerPlay reports for Tasks cube
Number of tasks handled by Agent Group for all media types
Average talk time for each Agent for single task contacts by Media type
Average Talk Time for each task type by media type
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Email specific Impromptu reports
Detailed Email Agent Response by Time Range
Detailed Email Agent Response by Date Range
Detailed Customer Email Management Tracking
Detailed Pool Response by Time Range
Email specific PowerPlay reports from Tasks cube
Detailed Agent Response by Time Range
Detailed Agent by Pool
Detailed Agent Interactions by Contact Entry Point
Detailed Contact Entry Point Traffic Trends
Pool Monthly Traffic Trends
Pool Traffic By Hour
Mail Account Summary
CMS specific reports (from the CMS cube)
Number of Calls by Time of Day and Skill
Number of Calls by Time of Day and VDN
Percentage of Calls by Time of Day and Skill for all ACDs
Percentage of Calls by Time of Day and VDN for all ACDs
Percentage of Calls by Time of Day and Skill for an ACD
Percentage of Calls by Time of Day and VDN for an ACD
Number of Abandons by Time of Day and Skill
Number of Abandons by Time of Day and VDN
Percentage of Abandons by Time of Day and Skill
Percentage of Abandons by Time of Day and VDN
Number of Abandons by Skill
Average Wait Time by Skill
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Contact Segment Impromptu reports (from the mma cube)
Number of Contacts by Site and by Time of Day
Number of Contacts by Site and by Day of Week
Number of Contacts by Site and by Day of Year
Number of Contacts by Site and by Week of Year
Number of Contacts by Site and by Month
Number of Contacts by Site and by Month of Year
Number of Contacts by Site and by Quarter
Number of Contacts by Site and by Year
Number of Contacts by Media Type and by Time of Day
Number of Contacts by Media Type and by Day of Week
Number of Contacts by Media Type and by Day of Year
Number of Contacts by Media Type and by Week of Year
Number of Contacts by Media Type and by Month
Number of Contacts by Media Type and by Month of Year
Number of Contacts by Media Type and by Quarter
Number of Contacts by Media Type and by Year
Number of Contacts by Site and Media Type
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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.
Avaya Response:
Comply. Both the Avaya Call Management System (CMS) and Avaya Operational
Analyst provide graphical reports as standard. Full listings of reports from Avaya
CMS are listed below and sample graphical reports are provided. Refer to Response
8.2.1 above for a list and description of Avaya Operational Analysts graphical
reporting capabilities.
Avaya Call Management System
The following table lists the Supervisor reports that are available. The reports you
see depend on your switch type, permissions, and system performance. CMS
Graphical Reports are indicated in the report name.
CMS Report name
RealTime
Historical
Outbound Split/Skill
x
Agent Attendance
x
Agent AUX
x
Agent Event Count
x
Agent Graphical Information
x
Agent Graphical Time Spent
x
Agent Group Attendance
x
Agent Group AUX
x
Agent Group Report
x
Agent Group Summary
x
Agent Inbound/Outbound
x
Agent Information
x
Agent Login/Logout (Skill)
x
Agent Login/Logout (Split)
x
Agent Report
x
Agent Split/Skill
x
Agent Status by Location
x
Agent Summary
x
Agent Trace
x
Busy Hour by Trunk Group
x
Busy Hour by VDN
x
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CMS Report name
RealTime
Historical
Call Record
x
Call Work Code
x
Event Count Summary
x
Graphical Active Agents
x
Graphical Allocated Agents
x
Graphical AUX Reserve1 Agents
x
Graphical AUX Reserve2 Agents
x
Graphical Average Positions Staffed
x
Graphical Busy/Abandon/ Disconnect
x
Graphical Maximum Delay
x
Graphical Multi-ACD Service Level Daily
x
Graphical Queue
x
Graphical Skill Overload
x
Graphical Split/Skill
x
Graphical Split/Skill Call Profile
x
Graphical Split/Skill View
x
Graphical Staffing Profile
x
Graphical VDN Call Profile
x
Multi-ACD
x
x
Multi-ACD by Split/Skill
x
Multi-ACD Call Flow by VDN
x
Multi-ACD Top Agent
x
Queue/Agent Status
x
Queue/Agent Summary
x
Queue/Top Agent Status
x
Reserve1 AUX Agents
x
Reserve2 AUX Agents
x
Skill AUX Report
x
Skill Status
x
Skill Top Agent Report
x
Split Status
x
Split/Skill Average Speed of Answer
x
Split/Skill by Location
x
Split/Skill Call Profile
x
Split/Skill Comparison
x
x
Split/Skill Graphical AUX Agents
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x
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CMS Report name
RealTime
Split/Skill Graphical AUX Top Agents
x
Split/Skill Graphical Call Profile
x
Split/Skill Graphical EWT
x
Split/Skill Graphical Service Level
Historical
Integrated
x
x
Split/Skill Graphical Status
x
Split/Skill Graphical Time Spent
x
Split/Skill Graphical Top Skill Status
x
Split/Skill Outbound
x
Split/Skill Queue
x
Split/Skill Report
x
x
Split/Skill Service Level
x
Split/Skill Status
x
Split/Skill Summary
x
System
x
System Multi-ACD
x
System Multi-ACD by Split/Skill
x
Top Agent Status
x
Trunk
x
Trunk Group
x
Trunk Group Summary
x
x
VDN Call Handling
x
VDN Call Profile
x
VDN Multi-ACD Flow
x
x
VDN Report
x
VDN Service Level
x
x
VDN Skill Preference
x
x
Vector
x
x
Work State Report for Reserve1Agents
x
Work State Report for Reserve2Agents
x
Special reports which focus specifically on the Average Speed of Answer are also
available such as the Split/Skill Graphical ASA (Average Speed of Answer) Report
which shows the average speed of answer for ACD calls answered in each selected
split/skill for each selected interval. The Split/Skill Graphical ASA (Average Speed of
Answer) Daily report shows the average speed of answer for ACD calls answered in
selected splits/skills for selected days.
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You can also see Average Speed of Answer on VDN based reports such as the VDN
Graphical Call Handling Report shown below. This report shows, for each VDN, the
cumulative number of calls that are answered, abandoned, and considered outflow
calls. The report also includes the average speed of answer.
This report shows how well the split or skill you specify performed compared to
your call center’s predefined service levels for the date you specify. This report has
four charts and displays a collection of split/skill call profile related data items at
the top of the report. A legend appears to the right of each chart.
The three-dimensional pie charts on the right side of the report show the
Percentage Answered Distribution (upper right quadrant) and the Percentage
Abandoned Distribution (lower right quadrant) for each service level increment. The
numerical value represented by each pie piece is shown inside the pie chart.
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The horizontal bar chart in the lower-left quadrant shows the actual number of ACD
calls answered within each service interval.
This report shows the percentage of ACD calls answered within the predefined
acceptable service level and the percentage of ACD calls abandoned for the date
and split or skill you specify.
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This report shows historical information and statistics for the specified agent. This
report is available in daily version only. Call center supervisors can use this report
to get an idea of how much time an agent spent on ACD calls, in available state, in
ACW, in AUX, and so on, for a particular day. This report enables the supervisor to
tell how much time the agent spent in AUX work state for each of the reason codes
defined for this Call Center.
8.2.2 Call-by-Call Reporting
The proposed solution must provide call-by-call reporting as an optional feature.
Vendor Response Requirement
Describe your system’s call-by-call reporting capabilities, if available.
Avaya Response:
Comply. Avaya has two offers for call-by-call reporting, as described below.
Avaya Operational Analyst
Avaya Operational Analyst (OA) is the optional reporting component for Avaya
Interaction Center, functioning as its operational data store and contact center
performance analysis system, provides the reporting and real time monitoring for
Interaction Center 7.1. It can also be used with Avaya CMS without Interaction
Center to receive and store External Call History Detail for one or more Avaya CMS
systems.
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Avaya OA will provide the following types of detailed call reports:
Work Item Detail Report (combined IC and CMS data) – This report provides
detail contact history information. From this report, you can drill down to
Segment Information and Wrap Up Codes reports. This reports helps answer
the question of what was the customer experience when contacting the
center, how many times on average was a work item of a particular type
handled, deferred or put on hold.
CMS Detail Report - This report provides detail contact history information
based on CMS call history detail records. From this report, you can drill down
to Segment Information and Call Work Codes reports. The reports act as a
hierarchy with Call Detail reports displaying multiple calls, each with a
corresponding Segment Information report. The Segment Information
reports list multiple call segments for a particular call – with each segment
having a corresponding Call Work Codes report.
For each customer interaction, the Interaction Center Engine captures all relevant
customer information in real time, as it occurs, across systems and locations. By
sharing customer information across systems, agents and communications
channels, companies can provide superior, consistent, and synchronized customer
service. The IC Engine creates a shared data object called the IC Electronic Data
Unit (EDU) for every interaction to record the cradle-to-grave history of that
interaction and allow each system and agent that interacts with that customer
access to the shared data.
By sharing customer information across systems, agents and communication
channels, companies can provide better-informed, consistent and synchronized
customer service. A shared data object, the EDU, is created for each interaction to
record the cradle-to-grave history of that interaction, and allow each system and
agent that interacts with this customer to access the shared data and contribute to
it. This allows all systems and agents to know what has occurred so far in each
customer interaction, independently of the media channels through which
interactions occur. Workflows executed by the Avaya IC Routing Service (IC
Workflow Server) use the EDU record information and then contribute more data to
it based on customer profile and other stored data. Agents receive “screen-pops” in
the Avaya Agent interface or directly in applications and access the EDU in a
consistent way regardless of the type of interaction.
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An EDU record is created for every interaction in the system, and remains until all
agents and servers have completed their activity related to the contact. The EDU
provides the data context for the work item, and contains several categories of data
elements:
Parameters from the network that identify the source and destination of the
work item (phone ANI & DNIS, email sender and recipient, etc.). This data is
always present and the format is fixed for a given channel.
Statistics data provided by the Channel Interface that measures the duration
of each activity related to the work item (queue time, talk time, hold time,
etc.). This data is always present and the format is fixed for a given channel.
Wrap-up information from each agent that participates in handling the work
item. This data is always present and the format is fixed.
Business-value data, such as data collected in the IVR or as part of the
segmentation and qualification workflow processes (customer ID, account
status, etc.). There may also be extended contact completion information,
such as business outcome of the interaction. These elements depend on
customization, and may or may not be present.
NICE Analyzer
NICE Analyzer provides access to detailed tracking information on each individual
call, from cradle (when the customer dials in) to grave (when the customer hangs
up). For example, detailed information might include a caller’s time in queue,
whether IVR was selected, the agent who handled the call, call transfer or hold
times, whether the caller abandoned, and so on.
NICE Analyzer also allows historical information to be stored for months, even years
after the actual call was received. (Note: Adequate disk space is required to support
the desired storage interval.)
Customer Experience Report (CER)
The NICE Analyzer Customer Experience Report (CER) presents a total view of a
call center interaction, regardless of the number of sites, call segments, or source
of the call.
Call Information displays total information about the call, including the start
and stop time, day of week, number of call segments, and total duration of
the call. Call information is presented once per CER display, while the
number of call segments varies.
Segment Information includes Calling Party (provided the switch is
configured for ISDN and ANI is provided from the network), DNIS digits,
Prompted Digits, Trunk Group, and length of the call segment.
Call Handling includes the Vector Directory Number, Vector Number, Vector
Treatment, Skill, Skill Level, Times Held, Total Hold Time, and Answering
Agent. After Call Work, Call Work Codes, and Stroke Counts also appear in
this section, as well as the segment or call disposition.
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Customer Experience Report
Optional Avaya Call Recording – NICE Call Logger System Integration
One of the most exciting features of NICE Analyzer is the integration of voice
recordings from the NICE Call Logger System into the NICE Analyzer Customer
Experience Report. By providing the NICE Call Logger System integration, all audio
files captured for a given contact, whether it occurs on one logger or on multiple
loggers, are displayed as an audio link within the Customer Experience Report
(CER). NICE Analyzer ties the data and audio file together using a unique identifier
created on the switch that, upon creation, follows both the data file and the audio
file for the life of the call. NICE Analyzer uses this field to tie the pieces together for
the most meaningful piece of information a management professional can possess,
the actual customer experience.
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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 selfservice interaction application.
Avaya Response:
Inbound Calling
Comply. The Avaya Interactive Response (IR) supports inbound calling both behind
and in front of the Avaya Call Center. Telephony interfaces provide the
telecommunications link between the PBX/Switching system (caller) and the Avaya
Interactive Response system (IVR application). Telephony links include:
• NMS AG4040/3200 – Quad Card
− T1 – 96 ports/board
− E1 – 120 port/board
− 75 Ω and 120 Ω
− SunFire V240 Dual CPU
• H.323 VOIP with Avaya Media Servers
− DEFINITY, Release 9.5 or greater
− MultiVantage 1.X, Avaya
Communication Manager
− Capacity of 240 ports
• NSM AG4000/1600 – Dual Card
− T1 – 48 ports/board
− E1 – 60 ports/board
− 75 Ω and 120 Ω
− SunFire V240 Single CPU only
Voice over IP Telephony with Avaya Media Servers
Voice over IP connectivity offers additional flexible deployment alternatives that
have not been available in any previous Avaya release. This offering allows ports on
Avaya IR to act as IP Stations on Communication Manager using CCMS/H.248
Signaling and leverage CM features, these IP channels will be configurable as
agents for further CTI oriented capabilities, however CTI is not an essential
requirement for call control for IR. Advantages:
o
More flexible, distributed architecture
o
Leverages existing IR servers and resources located anywhere on the
IP network
o
Helps reduce facilities/hardware capital and operational costs
o
Distributed systems offer greater resiliency and robustness of solution
o
H.323 connectivity to Avaya Communication Manager via G.711
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New Natural Micro-Systems (NMS) Telephony Connectivity Boards (E1/T1)
Digital connectivity is provided on industry-standard telephony boards. Commercial
AG4040 T1/E1 telephony cards for the Avaya IR system are provided by Natural
Microsystems (NMS). Digital connectivity offers speed and efficiency as well as call
progress tone detection. The Avaya IR on a Sun Fire V240 hardware platform can
have a maximum of 2 Quad T1/E1 cards. No NMS cards are used in a VoIP
configuration.
Each Quad NMS card provides 4 T1/4 E1 digital trunks that support the following:
G.711
Echo cancellation
Fax
On-board speech energy detection
Avaya Interactive Response will support standards based trunk-side and line-side
protocols as defined below. (Note that only T1 is offered within the U.S., Canada,
Hong Kong and Japan)
R2MFC
X
X
X
X
X
X
X
X
E1
QSIG
E1 CAS
X
X
X
E1 PRI
E1 Loop
X
X
X
T1 PRI
T1 Loop
T1 E&M
4 ESS (Central Office)
5ESS (Central Office)
Nortel (non Avaya PBX)
Siemens Hicom 300 (non Avaya PBX)
Avaya Communications Manager (G3R, Gsi,
etc)*
S8300/G700++
S8700 Media Server/(CM2.0)++
S8700 Media Server/(CM3.0)++
X
X
X
X
X
X
X
X
X**
X**
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
* Note:
G3R supports CM software up to CM1.3.
Si or CSI (ProLogix) supports CM software CM1.3, CM2.1, CM2.2, CM3.0, and
CM3.1.
** The avaya IR supports the QSIG protocol using NMS connectors that use
industry-wide QSIG standards. The Avaya IR was certified with the CM only but
because of the standards implemented within NMSQSIG should also integrate with
Nortel and Siemens Hicom switches.
++This release certified the Communications Manager using the S8700 server
against the Avaya IR. Other CM servers/gateways (S8100, S8500, S8300 with the
G350 gateway) have not been certified with the Avaya IR product. The following
represents the certified CM servers.
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Line-side protocol that conforms to the TIA/EIA-464B-1996 loop start (FXS)
protocol is included in the base Voice Channel License. Line-side protocol
provides digital emulation of analog lines, thus permitting flash transfers to
agents; generally supported by central office switches, PBXs and channel
bank under the name of Foreign eXchange Station Protocol, off-premise
station (OPS) protocol or line-side T1/E1. Avaya IR supports the Loop Start
Protocol (ANSI TIA/EIA-464B-1966 Section 6.2.3).
ISDN Trunk interface with enhanced message based signaling for
answer/disconnect supervision, dialed number (DNIS), calling party number
(ANI) and other information elements. The ISDN or DSS-1 protocols are
based on ITU Q.921 and Q.931 standards. Avaya IR supports ISDN/DSS-1
standards for:
o
AT&T PRI
o
Nortel PRI – Nortel A211-1
o
National ISDN – Telecordia (BellCore) SR-3875
o
ETSI PRI – ETS. Avaya Interactive Response will provide trunk-side
protocol support (EN 300 403-1, v1.2.2 – 1998 – 04) for ETSI-PRI for
the European ISDN market.
Messaging For Agents
Comply. The Avaya IR does support messaging applications such as callback
messaging which can speak the Expected Wait Time, offer callers the option to
leave a message or continue to wait in queue, and have the added capability of
queuing and delivering callback messages to agents and launching the callback at
the scheduled times or when agents become available. Callback messaging
applications for Avaya IR are available as custom or packaged applications from our
ISV partners or through Avaya CSI group.
The messaging solution proposed for agents is dependent upon your messaging
requirements. If you have advanced or high volume, full featured messaging
requirements, rather than an IVR based messaging solution, we recommend the
Avaya Modular Messaging solution which supports unified messaging capabilities at
the message storage level using an Avaya Message Server, Microsoft Exchange, or
IBM Lotus Domino. Modular Messaging is a powerful IP- and standards-based voice
and fax messaging platform designed for single- or multi-site global enterprises. It
offers exceptional scalability and a superior feature package of call answering and
messaging capabilities. Messages are accessible any time, anywhere from a wide
array of access devices including telephones, fax machines, or PC graphical user
interfaces. Voice, email, and fax messages can be stored in a single inbox either on
an Avaya Message Storage system or on your Microsoft Exchange or Lotus Domino
message storage servers. The Avaya Call Center can route calls to specific
mailboxes based upon a variety of call related information or current conditions in
the call center. Reporting is provided by the Modular Messaging application.
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Speech Recognition
Natural Language Speech Recognition
Avaya Interactive Response works in conjunction with leading speech recognition,
speaker verification and text to speech technologies to deliver speech enabled
service applications. The Avaya IR supports both legacy Avaya CONVERSANT®
applications and VoiceXML 2.0, both sets of applications can run concurrently on
the same system. The same VoiceXML interpreter is integrated into the Avaya Voice
Portal providing customers with investment protection and a clear migration path.
Avaya IR 2.0 provides support for WebSphere Voice Server 5.1 speech technologies
from IBM. This provides customers with the ability to have a complete end to end
solution with Avaya and IBM. Avaya IR 2.0 provides a standards based MRCP
interface to speech engines. This will provide customers will a standard way to
integrate to speech technology. Release 2.0 provides updated support for market
leading speech technologies including Nuance, OSR 3.0, Speechify 3.0, RealSpeak
4.0, OSDM 2.0. These speech offerings combine to provide support for 44 NLSR
languages and 20 TTS languages.
The following speech technologies are supported:
Nuance
o
OpenSpeech Recognizer 3.0, 2.0
o
Speechify TTS 3.0
o
RealSpeak 4.0, 3.5
o
Speech Secure 3.0
o
OpenSpeech DialogModules 3.0, 2.0
Nuance—TTS, NLSR, Verifier, Speech Objects
o
Recognizer 8.5, 8.0
o
Vocalizer TTS 4.0, 3.0
o
Verifier 3.5
IBM WebSphere Voice Server 5.1
Our distributed client/server NLSR architecture typically allows us to support more
NLSR ports per system than our NT competitors. Most of our competitors use an
“in-the-skins” or “all-eggs-in-one-basket” approach, running their NLSR software
natively on their IVR platform. NLSR is extremely CPU intensive and can easily
consume IVR resources when running natively on the box. A distributed
client/server approach allows us to easily and inexpensively scale by adding more
off-the-shelf platforms available from the open market.
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Text-To-Speech
Comply. With Proxy Text-to-Speech (PTTS), you can include speech in an
application using text as input. The text is converted to synthesized speech through
the third-party Text-to-Speech (TTS) engine. PTTS can be used for text that is
retrieved from a database or a host or for prompts. Text can be spoken in an
application with synthesized speech. PTTS is an alternative to using prerecorded
phrases for voice response. TTS can be essential in those applications that must
speak dynamic text (for example, names and addresses) or that have large
amounts of speakable text (for example, electronic news). Without TTS, these types
of applications can require many hours of recording and much disk space. These
applications can also use TTS for consistency in static text. The PTTS technology
can distinguish between different classes of text, such as zip codes and telephone
numbers, and pronounces the text string in the appropriate spoken format. When
constructing speech, parameters such as pitch and duration are adjusted to make
the outcome sound more natural. In addition, the ASCII text is pre-processed to
expand abbreviations. For example, "Dr." is expanded to "doctor" or "drive,"
depending on the context. Speech processing is done using one or more auxiliary
computers connected to the Avaya IR system in a client server configuration.
The following hardware is required for the PTTS feature:
Separate server for the TTS server software (speech engine).
LAN for connecting the Avaya IR system to the TTS server.
The following software is required for the PTTS feature:
Proxy Text-to-Speech package
Speech Proxy package
Vendor TTS server software (installed on a separate system). The PTTS
feature interfaces with the following third-party TTS engines:
o
ScanSoft’s SpeechWorks Speechify (http://www.nuance.com)
o
ScanSoft’s SpeechWorks RealSpeak (http://www.nuance.com)
o
Any system compliant with Speech Application Programming Interface
(SAPI) 4.0, including Loquendo TTS (http://www.loquendo.com). Note
that Loquendo TTS is supported for TAS applications only, not for
VoiceXML.
TDD
Comply. Although called a modem, the TDD modem might better be called a TDD
recognizer. This is a software package and not a hardware device that can be used
in IVR applications to allow communications with the hearing-impaired, in
compliance with Section 508 of the federal Rehabilitation Act of 1973. The TDD
modem acts much like a speech recognizer. It even uses the speech proxy server to
communicate with the Avaya IR server. When the IR system receives a TDD tone,
that tone is passed to the TDD modem. The TDD software then uses a grammarlike algorithm to determine the corresponding DTMF tone, which it then returns to
the IR system.
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Integration with a Customer Self-Service Interaction Application while
Maintaining Queue Position
Comply. The Converse vector command allows a caller to be connected to a vectorcontrolled split/skill, usually serving a voice response unit, while retaining its place
in queue for the primary split/skill. The integration is Inband and does not require
CTI integration. This feature allows voice response applications for the Avaya
Interactive Voice Response (IVR) system to make valuable use of caller wait time.
One of the strongest features of this voice response integration with the call center
is the ability to deliver self-service options to callers while waiting in queue for a
live agent. By providing the caller with useful options, the caller is better served,
and the call center manager can now manage peak queue volumes without hiring
additional expensive resources. Offer your callers a variety of customer self-service
options that make their wait time more productive. IVR applications include
information bulletin boards, audiotex, form filling, transaction processing, dynamic
announcements, expected wait time announcements, custom call routing, and
callback messaging as examples.
Avaya Communication Manager can use the Converse vector command to pass
information to and from a voice response unit (VRU) such as the Avaya Interactive
Voice Response (IVR) system. The data passed (such as ANI, DNIS, expected wait
time, queue position, or digits) is used to perform database lookups or execute IVR
scripts in order to determine a Route-to destination which can be passed back to
the Avaya Media Server or Avaya DEFINITY® Server to support Custom Call
Routing applications.
CTI and Call Control Services
Comply. Data-related Interfaces enable information to be passed between the
Avaya Interactive Response (IR) System and databases co-located on the Avaya IR
or to other downstream systems via LAN connections. To make these connections,
the Avaya IR system accommodates:
Computer Telephony Interface (CTI), such as JTAPI over Ethernet, enables
an Adjunct/Switch Application Interface (ASAI) connection between an Avaya
Media Server or DEFINITY® server and the Avaya Interactive Response
system over the Ethernet interface. ASAI supports advanced intelligent call
routing and screen-pop applications and provides dynamic port allocation and
script triggering.
Application Enablement Services delivers the CTI architecture and platform
that supports existing Call Center application requirements, along with the
new emerging applications programming interfaces (APIs), as they become
available. It consolidates multiple CTI server platforms onto a single server
while supporting the leading industry APIs including TSAPI, JTAPI, Avaya
CallVisor LAN (CVLAN) API, Definity LAN Gateway (DLG) and the Avaya
Communication Manager API (CMAPI) now called Device, Media and Call
Control (DMCC). This platform provides complete backwards compatibility for
all these APIs ensuring that the Application Enablement Services platform will
serve legacy, current and future application needs.
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TN3270E is the preferred method for connecting Avaya Interactive Response
system to a host computer. TN3270E provides 3270 sessions directly to the host
over Ethernet, TCP/IP connectivity.
Adjunct/Switch Applications Interface (ASAI) feature
The ASAI feature is a digital signaling interface that provides a LAN or VoIP
interface between an Avaya Communication Manager system and adjuncts. The
ASAI feature:
Routes calls to agents based on information from a database.
Delivers account information or caller profiles to the agent terminal at the
same time as the call.
Provides the ability to adjust application parameters.
With ASAI, the voice system can monitor and route calls on the switch. This
interface operates over an Ethernet TCP/IP link connected to MAPD in CVLAN mode.
When the ASAI interface is used in conjunction with digital line side T1 or E1 loop
start interfaces or VoIP, the voice system can monitor and control incoming calls. It
also supports access to ANI and DNIS and supports ASAI transfer, which is faster
and more reliable than a flash transfer. The ASAI feature includes the following
capabilities:
Capability
Description
Universal Call UCID provides a unique identifier (8-byte binary or 20-character
ID (UCID)
ASCII) for every call in an Avaya Communication Manager call center
customer environment for uniform data-tracking for all call-related
data in a call center, regardless of the system. Avaya
Communication Manager uses the ASAI interface to pass the UCID to
adjuncts.
ANI
Information
Indicator
(ANIII)
ANI-II provides a number that indicates the class of service of the
customer who is calling, such as residential, coin, or wireless.
User-to-User
Information
element
(UUI)
With UUI, the customer can specify additional information to be
passed in external function arguments, which can contain up to 96
bytes of information.
CallVisor
CallVisor libraries are supported over a TCP/IP stack on an Ethernet
LAN.
The full CallVisor CVLAN client of ASAI interface software is also provided with the
ASAI feature package to facilitate building ASAI applications in C code. Avaya
Professional Services provides development expertise in ASAI and the system, and
other independent software vendors (ISVs) can develop custom applications using
the ASAI API, thereby providing the optimum solution when you require full ASAI
integration with the application.
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8.3.1 Script Development
Vendor Response Requirement
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.
Avaya Response:
Avaya Dialog Designer
Dialog Designer is an open-standards based Integrated Development Environment
(IDE) for voice self-service applications. Based on the widely accepted Eclipse.org
development framework, Dialog Designer is a drag-and-drop environment for
development and maintenance of speech and touch-tone applications.
Dialog Designer allows complete lifecycle activities associated with application
development including: design, integration, simulation, debugging, scripting, and
deployment. By leveraging existing web server environments for deployment,
customers can reuse existing web-based integrations, web services and database
assets and skills, and web application development tools to drive faster time to
market and reduced cost of ownership.
Dialog Designer is available to purchase for the cost of media and is licensed for no
cost as an included component of both the new Avaya Voice Portal and Avaya
Interactive Response. Dialog Designer features a multi-lingual application model
and pre-built templates for common self service actions that integrate with Voice
Portal and Interactive Response through the common VoiceXML 2.0 certified
browser.
Next generation Web Services interfaces are supported completely from definition
to access through support of the standard Web Services Description Language
(WSDL) and the Simple Object Access Protocol (SOAP/XML). These interfaces,
along with additional Java2 Enterprise Edition support (J2EE), further speed
development and significantly reduce integration time and cost.
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Live Highlighting and
Debugging
Active Voice
Browser
Tomcat Console
Slide 21
Avaya – Proprietary (Restricted) Solely for authorized persons having a need to know pursuant to Company instructions
One of the strongest features of Avaya Dialog Designer is the capability Dialog
Designer offers for testing and debugging your speech application projects. In
Dialog Designer, you can simulate and test virtually every aspect of your speech
application project, including its response to error conditions. The Voice Portal
Management System also logs and identifies errors encountered with the
application.
In Dialog Designer, you can simulate the following features and functions of a call
flow:
The calling number (ANI)
The called number (DNIS)
DTMF inputs
ASR inputs, either using a microphone or by typing in a response
Problems with input recognition, such as No Match and No Input conditions
Caller hang ups during the call
Call transfers and all possible transfer results
Passing variable values from one module to another, when you are testing
individual modules
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Confidence level of ASR recognition
Interaction Center (IC) and Computer Telephony Integration
connectors, using a connector simulator and connector scripting
(CTI)
Dialog Designer provides the following features and tools designed to help you
debug applications:
Highlighting during simulation - While the application project is in simulation
mode, Dialog Designer displays highlights each node as the Avaya Voice
Browser (AVB) processes the nodes. This feature makes it easy to track
progress through the application as the simulation progresses.
Input tab, AVB progress display - During application simulation, the AVB
presents a step-by-step readout of what is happening as the simulation
progresses. The AVB displays this progress in a pane of the Input tab. Even
after the simulation ends, you can scroll back through this output to analyze
what happened during simulation.
The AVB Log tab - This item is similar to the previous item, but the
information that the AVB displays on this tab is much more detailed than in
the Input tab display. This information is a detailed transcript of the AVB
activity during call flow simulation.
Debug tracing to output in Console view or trace log file - If you have debug
tracing enabled, Dialog Designer directs the output of the debug tracing to
two destinations: the Console view display and the trace log file. This output
consists primarily of a transcript of the VoiceXML output that is generated
while the application is being run.
Application tracking node with debug options - Dialog Designer provides a
special node, the application Tracking node, which was designed to aid in
debugging applications. This node makes use of two palette options, the
Trace item and the Report item, to help with application tracking and
debugging.
Scripting of inputs and responses - For those situations where you do not
want or are not able to use the various built-in mechanisms for simulating
caller responses, you can create an XML script to tell the application how the
caller responds during simulation.
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Avaya IVR Designer 5.3 – VoiceXML Graphical Service Creation Tool
Avaya IVR Designer has been enhanced to generate code that is compliant with
VoiceXML 2.0. You can write VoiceXML by hand, use emerging tools, or use Avaya
IVR Designer and continue to use your application developers for maintaining past
and future applications with one tool.
Current VoiceXML editing tools on the market "assist" the user in entering raw
VoiceXML script by color-coding the syntax of the scripting language as reminders
that those items need to be modified. Avaya IVR Designer is the first tool that truly
provides a graphical abstraction of the application call flow logic. The user does not
have to know the syntax of the scripting language in order to write applications;
however, if they want that level of control, Avaya IVR Designer does provide direct
in-line code entry of raw VoiceXML script.
The advantages of Avaya IVR Designer are:
Familiar “drag and drop” development model
No prior knowledge of VoiceXML development required
Supports native script or VoiceXML code generation
Can hand edit raw VoiceXML code with any editor
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Native VoiceXML Code Generation
Testing and Deploying Applications with IVR Designer
IVR Designer includes a set of tools that you use to design, edit, test, simulate,
debug, generate, transfer, and install the applications. Once you have created,
tested, and debugged your application, you can use the Code Generation /
Application Transfer tool to generate the code for the target system.
You can use these tools to diagnose and debug your applications as described
below:
Verify Design
The Verify Design tool allows you to check your application design for possible
errors or omissions. When you open the Verify Design tool, it automatically
searches through each call flow of your application. The results are displayed in the
Verification Results window.
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Simulation Tool
To aid the developer in refining the logic flow of their application, Avaya IVR
Designer provides a simulation tool that supports entry of digits, display of timeout
countdowns and node execution. The tool also supports starting at any node in the
call flow and single-stepping through the application.
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.
Avaya Response:
Comply. Avaya includes basic integrated forecasting capabilities (described below)
as standard on the Avaya Call Management System (CMS). For more advanced
scheduling and adherence functionality, Avaya recommends products from our
DevConnect (Developer Connection) partner Blue Pumpkin. The Blue Pumpkin
Workforce Optimization Suite features:
Blue Pumpkin Planner - Long-term strategic resource planning
Blue Pumpkin Director - Resource & Skill Deployment
Blue Pumpkin Activity Manager - Time & Activity Management
Blue Pumpkin Advisor - Performance Management
The Avaya Call Center integrates with Blue Pumpkin and all major workforce
management vendors through custom reports from the Avaya CMS system.
Avaya Call Management System (CMS) Forecasting
Forecasting is a software feature included within the Avaya Call Management
System (CMS) software. The Forecasting software package allows you to set up and
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run split forecasts, quick forecasts, and trunk performance reports. The period of
time new data is available for forecasting is a design parameter. The defaults are:
28 days for intra-hour split data, 392 days for daily split data, 365 days for special
days data, 31 days for current day data, and 399 days for intra-hour and daily
trunk group data.
The standard customization capabilities of CMS extend to forecasting. Forecast
reports can be customized, and forecasting data can be combined with other data
(such as exceptions, historical, or financial) on a single report.
You can use CMS Forecasting to do the following:
Provide you with the estimated number of agents and the number of trunks
required for each intra-hour interval.
Set objectives for Automatic Call Distribution (ACD) activities involving
agents, split(s)/skill(s), and trunk/trunk groups for which you want to get
forecasting information about upcoming dates or time periods.
Generate reports using historical data that predict call volume and agent
requirements for:
o
Today (Current Day report)
o
Any day up to 35 days in the future (Longterm report)
o
A given profit margin plus how many calls you can expect for a
split/skill, and how many agents you will need to handle those calls for
any day up to 35 days in the future (Financial report)
o
The remaining part of the current day (Intra-day report)
o
A special day, for example, a holiday or a special promotion day
(Special Days report)
Generate reports that provide information about agent positions required,
trunks required, and trunk performance.
Predict the staffing requirements of your call center in hypothetical
situations.
Increase profits by predicting when to reduce surplus labor.
Generate reports that justify increased staffing based on objectives and
predicted call volume.
Perform careful scheduling and planning to optimize productivity.
Estimate how many calls a given number of agents can handle per intra-hour
interval.
Estimate how many calls can be carried by a given number of trunks per
intra-hour interval.
Estimate a margin for predicting the difference between call revenue and call
costs for each intra-hour interval.
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Gather forecast calls carried information for each intra-hour interval in
Special Day reports.
The standard report types are described below.
Call Volume/Agent Forecast: Accurate Predictions
A call volume/agent forecast predicts the number of calls a split/skill will receive
(forecast calls carried) and how many agents will be required to handle those calls
(number of agents required). For Expert Agent Selection, this is the number of
agents which should have this number as their top skill. The types of call
volume/agent forecasts available are as follows:
Current Day Forecast is a forecast for today, based on historical data.
Longterm Forecast is a forecast for tomorrow or a day up to 35 days in the
future, based on historical data.
Special Day Forecast is a forecast for a day that has unique characteristics,
based on historical data.
Intra-day Forecast is a forecast for the remainder of today, based on
historical data and on data from the beginning of today.
Longterm Financial Forecast is a Long-term Forecast that has an additional
forecast of profit margins.
Hypothetical Forecast is a forecast for tomorrow or a day up to 35 days in
the future, based on hypothetical data defined by the user.
Hypothetical Financial Forecast is a hypothetical forecast that has additional
forecast or profit margins.
Quick Requirement Forecast Reports: Immediate Reporting
The following two quick requirement forecast reports can tell you, given specific call
handling objectives, how many calls you can handle at increasing resource levels.
Requirement forecast reports do not require set up and do not use historical data.
Therefore, you can begin using these reports immediately.
Agent Positions Required Report provides a table showing how many calls a
specified number of agents can handle given specified call handling
objectives. This report forecasts the number of agents a split/skill will need
as the number of calls increases.
Trunks Required Report provides a table showing how many calls can be
carried by a specified number of trunks given a specified trunk blocking
objective. This report forecasts the number of trunks a trunk group will need
as the number of calls increases.
Trunk Performance Report
The Trunk Performance Report tells, based on the number of calls in a period of
time in the past (usually a month), how many trunks in a trunk group will maximize
call handling efficiency. The Trunk Performance report estimates, at the busiest
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intervals in a specified range of historical dates, what the usage rate and blocking
percentages were for the selected trunk group(s). The report also tells you, given
the objective blocking percentage(s) specified in the Trunk Group Profile window,
how many trunks the trunk group(s) would have needed during the busiest
intervals to meet the objectives. This report does not actually predict the number of
trunks needed.
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.
Avaya Response:
Comply. Avaya has two offers for the VoiceCon Email Channel which can meet the
specifications above. Both of these offers are described below. Avaya offers Contact
Center Express to support the email channel for our mid-size contact centers
requiring standard Email Header Analysis, queuing and automatic delivery of Emails
as work items to contact center agents, and basic send/forward/reply email
features.
If Email Full Text Content Analysis, Approval Routing, and integration of a
Common/Canned Response database are required, Avaya Interaction Center is
recommended. Avaya Interaction Center supports our largest call center customers
while Avaya Contact Center Express is aimed at mid-sized requirements and
budgets. While targeted at the mid-markets contact center (150 agent desktop
range), this is not a capacity limitation and Avaya Contact Center Express can
continue scale to support your business well beyond your initial investment.
Avaya Contact Center Express – Email Channel
Avaya Contact Center Express manages the collection, queuing, and delivery of
voice and non-voice work items such as e-mail and chat sessions to an
appropriately skilled agent. Contact Center Express utilizes the powerful routing
algorithms resident in Avaya Communication Manager to determine the right
resource for the right interaction.
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Avaya Contact Center Express provides a set of multi-channel capabilities that
medium-sized contact centers can leverage and build upon:
Desktop applications, including Agent Applications, Supervisor Applications,
and Utility Applications. These out-of-the-box applications allow you to begin
working with new technologies within hours.
Framework applications for the contact center, including Intelligent Routing,
Interaction Data, and Centralized Configuration.
Multi-channel routing for voice, e-mail, and Web chat allowing you to create
true universal agents.
Outbound dialing with automated and agent-initiated Preview Contact.
Simple but effective, designed to solve costly outbound dialing issues, from
callbacks to targeted campaigns.
Powerful application development tools for complete customization and
integration capabilities.
Simple and fast wizards for desktop screen pops and routing rules.
Contact Center Express provides functionality that can easily and quickly adapt to
business dynamics without requiring a large budget and IT staff. Contact Center
Express is able to fully leverage the unique abilities of Avaya Communication
Manager, and provides multi-channel and agent performance enhancement
capabilities that translate into real results for your contact center.
The Avaya Contact Center Express Media Director distributes non-voice work items
to contact center agents. This item could be an email, a web chat session or an
outbound call request. The distribution of the work item is achieved using the
queuing algorithms built into the Avaya Communication Manager server.
Non-voice work items originate from plug-in modules called media stores. Media
stores connect to disparate sources such as email servers or web servers and
interact with the Media Director and clients using a well-defined protocol. When a
media store receives a new work item from a media source (e.g., email server for
the Email Media Store, web chat for the Simple Messaging Media Store, or SQL
database for the Preview Contact Media Store), it creates a work item object and
passes a reference for that object to the Media Director. The reference tells the
Media Director what queue (queue ID) the work item is to be associated with and
what priority it must have in the queue. Using the information in its configuration
that relates specifically to that queue, the Media Director asks the Avaya
Communication Manager server (Application Enablement Services Server (AE
Services)) to queue it to the appropriate skill group.
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When an agent logged into that skill group becomes available, the Avaya
Communication Manager server delivers the most appropriate work item to the
agent. The Media Director is monitoring the Vector Directory Number (VDN) and
sees the work item delivered to the agent. The Media Director transfers the work
item reference with the oldest, highest-priority (1-10 priority levels) object to the
Media Proxy. The Media Proxy delivers the reference to the correct client application
based on the specified work item type. The client application uses the reference to
retrieve the data directly from the actual work item at the media store.
The Email Media Store allows you to blend customer email inquiries with inbound
telephone calls, essentially using this work to fill in the gaps between peaks in
inbound call traffic.
The Email Media Store receives emails from one or more mail servers using the
POP3 protocol. Installed on a Microsoft SQL server, it uses its configuration data
and the information specified in the database schema, to:
distribute emails sent to certain mailboxes to certain queues in the Media
Director
manage that distribution by making email queues 'open' for certain times and
days of the week
give queuing priority to emails received from special customers
assign different queuing priorities to the first email a customer sends and all
subsequent emails they send as part of the same conversation
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reject emails from certain customers and automatically email them that this
has happened only allow emails from certain customers to queue to a certain
email queue automatically inform a customer (via email) that their email has
been received during or outside the operating hours of that queue.
Every period, the email media store connects with the specified mail server and
downloads new email items. Email from one mailbox is matched with one email
queue in the media store and each email queue has a priority in which to send
email to a certain Media Director email queue. Once downloaded, the email in the
mail server is deleted, the connection closed, and a series of processing steps
occurs:
Check for automated responses/error messages
Check for allowed and denied senders
Interrogate the email header for an existing conversation ID
Check for priority customer status
Contact Center Express also supports routing between autonomous contact centers.
With the power of Avaya call center routing capabilities, contact data can be
transferred along with the call or non-real time work item across the WAN and
seamless integration of contact centers supported.
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Supported Platforms/Minimum System Requirements
Telephony
Software
Hardware
Switches
CTI
Avaya DEFINITY® G3V8.3, G3V9,
G3V10; Avaya Communication Manager
1.2, 1.3, 2.0, 2.1, 2.2, 3.0, 3.1
Any switch hardware and media
gateway supported by Avaya
Application Enablement Services
Switches
Multi-Site
Avaya DEFINITY® G3V8.3, G3V9,
G3V10; Avaya Communication Manager
1.2, 1.3, 2.0, 2.1, 2.2, 3.0, 3.1
Any switch hardware and media
gateway supported by Avaya
Application Enablement Services
Contact Center Express Platforms
Switches
See detail above
CTI
Avaya Application Enablement Services 3.0, 3.1
Reporting
Avaya CMS R9, R11, R12, R13, R13.1
CMS Supervisor releases compatible with the CMS platform.
Note: CMS Supervisor has been tested on Windows NT 4.0 and has
no known issues; however, Supervisor on NT 4.0 is permissive use
only since Microsoft no longer supports Windows NT 4.0.
BCMR Desktop Release 2v3
IVRs
Avaya Conversant v7, v8 and v9;
Avaya IR (Interactive Response) 1.0, 1.2, 1.2.1, 1.3, 2.0
Avaya Voice Portal 3.0
Servers OS
Microsoft Windows NT 4.0 SP6a (permissive use since no longer
supported by MS)
Microsoft Windows 2000 Server SP4;
Microsoft Windows 2003
Databases
Microsoft SQL Server 2000 SP3 (on supported Server OSs)
(Microsoft SQL Server 2000 is supported on Windows NT 4.0)
Development and
Design Tools
Microsoft Windows 98
Microsoft Windows NT 4.0 SP6a (permissive use since no longer
supported by MS)
Microsoft Windows 2000 Professional SP4
Microsoft Windows XP Professional
Microsoft Windows 2003 Professional
Visual Studios
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Contact Center Express Platforms
Administration OS
Microsoft Windows 98
Microsoft Windows NT 4.0 SP6a (permissive use since no longer
supported by MS)
Microsoft Windows 2000 Professional SP4
Microsoft Windows XP Professional
Microsoft Windows 2003 Professional
Microsoft Management Console (MMC) 1.1 or later
Agent Desktop OS
Microsoft Windows 98
Microsoft Windows NT 4.0 SP6a (permissive use since no longer
supported by MS)
Microsoft Windows 2000 Professional SP4
Microsoft Windows XP Professional
Email Servers
Microsoft Exchange 5.5 and 2000 (on supported Server OS)
Note: Only POP3 is supported
Application Enablement Services server, with TSAPI basic and Advanced
licenses.
Minimum of additional 128 MB RAM for use of license server recommended
Server requirements
• 2.4 GHz Pentium IV or higher processor
• 1 Gb of RAM memory (minimum)
• 40Gb hard drive (minimum)
• A CD-ROM drive, software may be installed from a server
via the network.
A mouse compatible with the supported Windows operating systems.
56 Kbps modem
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Client PC requirements
Microsoft Windows 98 Second Edition, Me, NT 4.0 (with Service Pack 6),
2000 or XP Professional in XP or Classic styles.
A 266 MHz Pentium or higher processor.
At least 32MB of RAM memory (64MB for Windows NT 4.0, 2000 or XP
Professional).
50MB of free hard disk space for the application, 10 -18MB for online
documentation (file sizes depend on the language you install), and around
2MB for client sample applications.
A CD-ROM drive.
A SVGA or better video controller, and monitor with resolution set at 800 x
600 pixels or higher.
A mouse or other Windows-compatible pointing device.
A TCP/IP LAN connection to the Telephony Server.
TSAPI client software (Embedded into the CCE agent desktop)
Avaya Interaction Center (IC) Email Channel
IC Email provides the ability for agents to receive and respond to email so
organizations can promptly and efficiently manage increasing email loads without
having to increase the number of contact center agent resources at the same rate.
Email contacts are routed, blended, and tracked at every step by the IC Engine.
Email Automation is provided to analyze message content, determine the nature of
the customer’s request, and make a decision about whether to automatically
respond or to provide a suggested response to the agent with the queued email.
Responses can be automatically generated and populated with customer specific
data accessed from the Customer Interaction Repository or from external data
sources.
This email capability provides the contact center the ability to handle emails
blended in with other contacts such as inbound calls and web requests in common
way using a common desktop interface, business rules and reporting.
Business rules can be created for routing and responding to customer email based
on customer value, topic of the email, and conditions in the contact center. These
emails can then be delivered to an agent with a customer context and the tools for
the agent to respond to that email. Depending on the business logic, certain emails
can be responded to automatically without involving an agent.
For all email interactions, the original customer email, and all contact center
responses are stored in the customer repository and are available for review.
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Natural Language Content Analysis
Avaya Interaction Center 7.x email provides a powerful natural language content
analysis capability, known as Avaya Content Analyzer. Content Analysis is an
optional component that is priced separately. Content Analyzer provides the ability
to identify the language and the topic of an email. The results of content analysis
can then be used for the following functions as part of the email business rules:
Intelligent auto acknowledgement
Routing of the email based on topic and language.
Suggested responses for the agent
Automatic responses to email messages.
Quality assurance screening
Junk mail screening that is specific to the business rules
Extended Email Flows
Avaya Interaction Center provides for extended email flows for more than just “one
and done” capability of email handling. These capabilities are provided as out of the
box workflows, email states, and web agent user interface capability:
Email Analysis and Qualification – The email analysis and qualification
workflow provides the capability to analyze the email content, look up the
customer, based on these criteria, route the email appropriately, or autorespond.
Interim Responses – Avaya Interaction Center email provides the capability
for the agent to provide interim responses to the customer, without closing
the email. This can be used to provide partial responses to the customer,
provide status to the customer, and request additional information from the
customer. All of these interim responses are captured in the contact record.
Subject Matter Expert – Avaya Interaction Center email provides the
capability for the agent to forward the email to a subject matter expert either
inside or outside the contact center. The response from the subject matter
expert then comes back to the agent or the group (configurable) to be
integrated into a response to the customer. There is an alert mechanism that
is part of this feature that will set an alert to notify the agent if the reply
from the subject matter expert is not received within a defined time limit.
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Quality Assurance – Avaya Interaction Center 7.x provides the capability to
screen outbound email messages for quality assurance before it is sent to the
customer. This functionality is part of the outbound email workflow. There
are several mechanisms that can trigger sending an email to an approver.
o
Per agent quota – administrators can set a % sample rate on a per
agent basis.
o
Keyword Matching – Various defined keywords such as “guarantee”
o
Content Analysis – Various topics, or failure to match approved
responses.
In the case where the email is sent to the approver, the approver can review the
email, and either edit the email and send it, reject the email with comments back to
the agent, or send it out without changes.
Email Interfaces
Avaya Interaction Center Email interfaces with the customer’s email server via a
POP-3/SMTP interface. This interface is supported by most leading email services,
including Microsoft Exchange. The email is polled from the customer provided email
server, and stored and delivered in Avaya Interaction Center. The email is delivered
to the agent in the Avaya Web Agent user interface.
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.
Avaya Response:
Comply. Avaya has two offers for the VoiceCon Web Center. Contact Center Express
supports live web chat from website requests and can collaborate in terms of
responding to real time information requests via chat. If your contact center
requires form sharing, co-browsing, Voice over IP (“Click here to talk to a live
agent”), and other advanced features described below, Avaya Interaction Center is
recommended.
Contact Center Express
Our Contact Center Express offer for mid-sized call centers supports a Simple
Messaging Media Store that sits between the Media Director and Contact Center
Express simple messaging gateways, such as the Web Chat Gateway. It provides
the base (common) messaging functionality required by these gateways, allowing
you to blend customer text-based messages (information requests) with inbound
telephone calls.
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Installed on a Microsoft SQL server, the Simple Messaging Media Store uses its
configuration data and the information specified in the database schema, to:
send simple messages from different gateways to different Media Director
queues
give queuing priority to messages received from special customers
reject messages from certain customers and automatically email them that
this has happened
only allow messages from certain customers to queue to a certain Media
Web Chat Gateway
The Web Chat gateway is the first simple messaging gateway provided with Contact
Center Express. It allows an agent and customer to have “I say/You say”
capabilities across the Internet. An application provided with CCE must be installed
on the customer’s web server to enable this capability.
Avaya Interaction Center Web Channel
The Avaya IC Web Channel allows the end user who is surfing an Internet site to
obtain help in a number of different ways. First, there is a self-help capability,
which allows the end user to search a self-help knowledge base and potentially find
answers to their questions. If the end user is not able to find an answer using the
self-help facility, they are able to escalate their request into the contact center for
assistance by live agents. When this contact is escalated, the end user has a choice
of media to use in interacting with the contact center. These choices are detailed
below.
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Web Self-Help, DataWake® and Escalation Features
Avaya Interaction Center Web channel provides a set of web site capabilities to
provide web customers with the ability to search frequently asked questions using a
web search engine. This provides a mechanism for customers to help themselves.
The web customer’s trail though the web site and self-help knowledgebase can be
tracked via the DataWake capability.
If the web customer wants to escalate to a live agent they can be presented with
choice of media to interact with the contact center. This choice of media can be
configured on a per tenant basis, and can be controlled based on customer
entitlements or transaction value.
When the interaction request is escalated into the contract center, the information
surrounding that contact such as customer ID, DataWake record, escalation page,
media type, language, etc. are passed with the contact request into the IC
workflow. This information can be used for routing and prioritizing the web contact.
While the web customer waits in queue for an agent, IC can push messages and
web pages to the customer’s browser. The content of these messages and pages
can be determined via the IC workflow. The customer can be shown information
such as estimated wait time, frequently asked questions and infomercial in queue.
When the web contact is delivered to agent, the agent also gets the set of
information that came with the escalation, such as the customer information and
DataWake record. In addition the agent’s browser will be updated with the page the
customer was on when they requested help.
Multiple Media Interactions
Avaya Interaction Center provides the capability for the web customer to have a
choice of multiple media both individually and in combination. Thus the web
customer and a contact center agent can jointly fill in a web form at the same time
they are talking, via either a standard telephone, or via voice over IP directly over
the internet.
Text Chat
Text chat allows the web customer to interact with the contact center using
messages that are typed over the Internet between the customer and the agent.
This capability is always in conjunction with web collaboration. The web customer is
presented a chat interface in a browser window whose appearance can be
customized on a per tenant basis, and can be localized into the customer’s
language. The contact center agent has the ability to handle multiple simultaneous
text chats up to an administered limit. The contact center agent also has an
interface that will allow them to respond to callers with canned messages, and
canned web links (URL’s)
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Web Collaboration and Collaborative Form Filling
Whenever a text chat session is active, the customer and agent also have the
ability to do web collaboration via URL sharing. This capability also supports
collaborative web form filling, where changes made on an HTML form by either the
agent or the customer are also updated on the other’s browser.
Sharing of complex web pages including forms within frames.
Support for cookie sharing to allow for sharing of pages such as “MyYahoo”
where the content is defined by a combination of the URL and Cookie data
Support for “off domain browsing” where the agent can push pages that were
not from the original domain
A follow-me capability for the agent to lead the caller.
Text Chat and Collaboration Transcript
The session data (text chat and collaboration requests) is sent to the customer over
a channel that is encrypted with 128-bit encryption. This communication can be
tunneled on the caller side in cases where the caller is behind a firewall.
This session transcript is stored in the customer repository as part of the interaction
data for this web interaction. In addition this session transcript can be emailed to
the customer at the conclusion of the session.
The form of this transcript is XML data, with the presentation being controlled by a
style sheet.
Voice Chat
Voice chat refers to the ability to simultaneously have a web chat and collaboration
session combined with a voice conversation between the web customer and the
contact center agent(s). This interaction is treated as a single interaction.
Chat and Phone – Web Chat and Phone refers to the ability for the web customer to
be able chat and collaborate with a contact center agent(s) while simultaneously
talking over a telephone. This kind of interaction can be specified by the web
customer upon escalation to the contact center, or launched by the agent. The
phone part of this interaction is automatically launched and combined with the chat
interaction, and is treated as a single interaction.
Chat and Voice Over IP – Web Chat and Voice Over IP refers to the ability for the
web customer to be able to chat and collaborate with a contact center agent(s)
while simultaneously talking to that same agent via the internet though their
computer. For the agent, this call is handled though the same telephone that they
use to handle regular inbound phone calls. This kind of interaction can be specified
by web customer upon escalation to the contact center, or the agent can create and
add the VOIP call after the chat and collaboration session is established.
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Scheduled Web Callback
The scheduled web callback is a capability for a web customer to request a callback
from the contact center in a scheduled timeframe. Since this is a phone contact
only, there is no chat or collaboration associated with the callback. For these
contacts, the web customer specifies a Callback request window with the earliest
and latest acceptable times for the call.
Join Us
Join Us is the ability to add an additional external party to and existing web contact.
This can be anyone with access to the web site. An example of the use of this
functionality would be a customer adding on their spouse to be able to see the
same web pages and hear the same explanation.
Lotus Sametime Integration (app sharing, whiteboard)
The Avaya Interaction Center web channel supports integration with Lotus
Sametime. Integration with Lotus Sametime can be used to provide contact centers
with the ability to provide additional collaboration capabilities such as application
sharing, presentations and white boarding. This is a loose integration that will allow
the launching of a Lotus Sametime session from an existing chat and collaboration
session. It is the customer’s responsibility to purchase and install Lotus Sametime.
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.
Avaya Response:
Comply. The Avaya offers for outbound dialing are as follows:
Contact Center Express Preview Contact Outbound Dialing provides
automated and agent-initiated Preview Contact. Simple but effective,
designed to solve costly outbound dialing issues, from call backs to targeted
campaigns.
Interaction Center supports both the Avaya Proactive Contact System or
depending upon customer requirements, an integrated soft dialer to provide
full featured Proactive Contact.
Contact Center Express Preview Contact
The Preview Contact Media Store allows you to blend on-screen customer contact
prompts with inbound calls, essentially using this work to fill in the gaps between
peaks in inbound call traffic. Preview contact is defined as distributing a customer
record to an agent so that the agent can initiate contact with the customer by
phone.
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Installed on a Microsoft SQL server, the Preview Contact Media Store retrieves
contact details from a SQL database. The task to contact a group of contacts is
defined in the database as a campaign. The campaign is prescribed to start at a
certain date/time and run until another date/time. It can run over multiple time
periods and may be recursive (e.g. starting every Monday morning at 9:00).
Campaigns can be scheduled to coincide with:
different shifts
quieter times of the day (low-peak call times)
times of the day when it is easy to contact customers.
A campaign's configuration identifies which queue work items must queue to and
their priority within that queue. The Outbound Administrator is a standalone
application that allows the creation, deletion and modification of various campaign
data sets and the importing and exporting of contact data.
For hardware and software requirements for Contact Center Express, refer to
Response 8.5.0 above.
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The Avaya Proactive Contact System
Avaya Proactive Contact System with Avaya PG230 Proactive Contact Gateway is a
proactive contact management solution that includes the world’s most accurate
predictive dialer, call progress analysis tools, and robust dialing algorithms. The
PG230 next generation digital switch enables easier integration to the contact
center infrastructure but with a simplified hardware configuration.
Avaya Proactive Contact System with Avaya PG230 Proactive Contact Gateway
provides industry-leading call classification (up to 97.5% accuracy for call
detection). Its proprietary dialing algorithms optimize agent productivity.
Additionally, it:
Integrates easily into existing contact center environments.
Enables delivery of up to 130,000 calls per hour.
Its open architecture enables ease of integration and support with third-party
applications such as:
o
Interactive Response
o
Quality Monitoring
o
CRM Applications
The Avaya Proactive Contact System allows business to efficiently interact with
their customers based on real-time performance information, the Avaya PCS
predicts when a contact center agent will be available to speak with a customer.
The system can be used to blend and manage agent resources in response to both
inbound and outbound customer calls. As a result, agents are kept fully productive
while the system ensures that no calls are abandoned because of unavailable
agents. Built to include enhanced predictive algorithms, The Avaya PCS virtually
eliminates abandoned calls by effectively connecting agents and customers.
The Avaya Proactive Contact System delivers a scalable solution with support for
1728 agents from single or multiple locations featuring:
State-of-the-Art Dialing Algorithms – Expert Calling Ratio® analyzes call
statistics every few minutes. It predicts dialing outcome probability and
averages talk and update times. It calculates the quantity of dialing attempts
to yield a stream of live connects–at a pace you control. The system can
place up to 130,000 calls per hour.
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Cruise Control -Cruise control automatically maintains the service level of
outbound dialing during a job and connects the calls to agents within a
specified period of time. During the job, you do not have to monitor or
modify the call pacing settings.
When you set up an outbound job that uses Cruise Control, you must define
the Desired service level and the Time to connect tolerance settings.
The system uses these settings to do the following:
•
Predict when to automatically dial phone numbers
•
Distribute phone calls within the tolerable time period that you set
Once you start a job that uses Cruise Control, you do not have to change the
settings. If you want to change the settings, you must stop the job. To
resume calling activities with the new settings, restart the job.
Call Blending – As inbound volume increases, a sophisticated call-blending
engine transfers calls to the blended inbound/outbound team when
necessary. Call blending minimizes sporadic inbound overloads and reduces
agents' idle time. Choose from two blending strategies: overflow-based, or
based on predictive analysis of inbound calling trends.
Industry-Leading Voice Detection – The Avaya Proactive Contact System
voice detection maximizes live-voice connects eliminating up to 97.5% of
busy signals, answering machines, voice mail, unanswered calls, pagers, fax
machines, modems, and operator intercepts. The system is up to 25% more
accurate than other Proactive Contact systems.
Superior System Management – Avaya PC Supervisor® is a powerful
supervisory tool that gives call center managers real-time information about
campaign and agent performance. It enables supervisors to set targeted and
effective campaign strategies, and provides status and agent activity reports
at any stage in the campaign. Supervisor features a graphical Microsoft
Windows-based interface with easy-to-use pull-down menus and input fields.
Multidialer Capabilities – You can create and manage log-ins and passwords
for multiple dialers from a single system, combine real-time data from
multiple dialers, and share user defined views across the enterprise. One or
more Avaya Proactive Contact Systems can run jobs using a single master
list residing on any other dialer.
The following optional application software is available for use with the Avaya
Proactive Contact System
Avaya Proactive Contact System Supervisor Modules
Campaign Editor: Campaign Editor features an easy-to-use, Windows-based
interface that enables supervisors to build and begin calling campaigns with
point-and-click simplicity. Supervisors can manage lists, create calling
strategies, select records, and design new campaigns—without extensive
computer training or experience.
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Monitor: Monitor provides a real-time view of system, job and agent
statistics. Supervisors can monitor as many of these activities (single or
group) as desired, including current call results, call quality, agent
productivity, talk times and progress toward campaign goals. Scope
Selectors present options for how much data can be viewed. Supervisors
can monitor inbound and outbound wait queues and view call completion
results for each campaign, thus analyzing campaign effectiveness.
Analyst: Analyst is a powerful query, reporting and analysis tool for your
Avaya Proactive Contact System. Analyst gives you the ability to assess
performance with real-time and historical data. In addition to over 18
comprehensive standard reports, Analyst puts ad-hoc reporting in the
hands of contact center management, helping to satisfy your needs for
timely, pertinent information on which to base operational and strategic
decisions.
Avaya Proactive Contact System Internet Monitor – This quality call center
campaign and agent-monitoring tool supports Internet technology in the call
center.
Avaya Proactive Contact System Administration Manager – Administration
Manager is a simple-to-use PC-based software tool that enables managers to
modify and maintain their Avaya Proactive Contact System.
Avaya Proactive Contact System Agent API – Agent API is a software
developer kit that helps customers build customized agent interfaces by
integrating data from their host computer and the Avaya Proactive Contact
System.
Avaya Proactive Contact System Scripting – With our call scripting program,
agents receive just-in-time access to the information they need—via
intuitive, easy-to-use, point-and-click scripts. This powerful scripting tool
controls the pace and content of every call. It allows database access,
providing critical information to agents when and where they need it.
The Contact Center World Leader
The Avaya Proactive Contact Solution is proven in more than 1,200 of the world’s
largest and most profitable contact centers, which together manage in excess of a
billion customer contacts annually. More than 80% of the Fortune 500® banking
and telecommunications companies use Avaya Proactive Contact solutions.
Avaya holds the #1 market share in the Proactive Contact Management solutions
and is well position globally to deliver a comprehensive solution in Outbound
Communications for the contact center and leverage the existing install base of
contact center customers to deliver a compelling solution that can optimize agent
efficiency and drive revenue or reduce cost for the contact center.
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Hardware and Software Requirements
The following requirements are necessary to support the Avaya Proactive Contact
System and optional software
Avaya Proactive Contact System Supervisor
Hardware
o
Intel Pentium 166 MHz PC
o
330 MB free disk space
o
32 MB RAM
o
Network interface card
o
Super VGA monitor (17" or larger)
o
SVGA accelerator card
o
CD-ROM drive
o
3.5" disk drive
o
Software
o
Microsoft Windows 98 or later, or Windows NT 4.0, SP3 or later,
Windows XP
o
Microsoft Winsock 1.1 TCP/IP stack
o
pcAnywhere 32, version 7.5 or later
o
Microsoft Office Professional for Windows 95 or later. If using a shared
database for more than three Supervisor workstations, a PC dedicated
to the data or a server for the data is recommended.
Avaya Proactive Contact System Internet Monitor
Web Server
o
A Web server (hardware and software) that allows the Avaya system
to Network File System (NFS) mount a home directory.
o
The Web server and Avaya system must be on a common network.
Avaya PCS Internet Monitor transfers approximately 40 KB of data
from each Avaya system to the Web server in 15-second intervals.
o
The server disk space Internet Monitor requires varies depending on
the size and number of Avaya systems. The following table outlines
the necessary server disk space. Each row of data indicates required
disk space for a single Avaya’s system.
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Web Browser
To view Avaya PCS Internet Monitor, use a Web browser that supports
frames and client pull, such as Netscape Navigator 2.0 or later, or
Microsoft Internet Explorer 3.0 or later.
Avaya Proactive Contact System Administration Manager
o
Hardware*
o
Intel Pentium 166 MHz PC
o
16 MB RAM for Windows 98, or 32 MB for Windows NT
o
100 MB free disk space
o
CD-ROM drive, 4x minimum, or 3.5" disk drive
o
Network interface card: token ring 4/16 MB per second or Ethernet
10/100TX
o
SVGA accelerator card with 1 MB VRAM
o
Super VGA 17" monitor
o
Sound Blaster 16 Sound Card with compatible speakers and
microphone
o
33.6 K baud data modem and DID line - Required unless PC can be
accessed through a TCP/IP connection for remote support software.
* Minimum hardware to run Administration Manager and Supervisor on the
same PC requires 32 MB RAM and 320 MB of available hard disk space.
Software
Microsoft Windows 95 OSR2 and Y2K Update Patch or Windows NT 4.0 SP5
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.
Avaya Response:
Avaya Application Enablement Services provides an enhanced set of Application
Programming Interfaces (APIs), protocols and web services that expose the
functionality of Avaya communication solutions to corporate application developers,
3rd party Independent Software Vendors (ISVs) and system integrators. This open
standards-based solution runs on a Linux server and is tightly integrated with
Avaya Communication Manager and Avaya Contact Center solutions. Application
Enablement Services provides a new, open platform for supporting existing
applications, and will be the catalyst for creating the next generation of applications
and business solutions for our customers.
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Application Enablement Services provides the ability for traditional IT data
application developers to interface to Avaya Communication Manager through
standard Web Services via SOAP/XML methods. This enables integration of business
and communication applications to leverage the power of real time telephony and
system management functions of Avaya Communication Manager. It provides these
functions via a Web Service, providing a standard and familiar method for IT data
application developers to implement new and innovative solutions.
Application Enablement Services delivers the CTI architecture and platform that
supports existing Call Center application requirements, along with the new
emerging applications programming interfaces (APIs), as they became available. It
consolidates multiple CTI server platforms onto a single server while supporting the
leading industry APIs including TSAPI, JTAPI, Avaya CallVisor LAN (CVLAN) API and
the Avaya Communication Manager API (CMAPI) now called Device, Media and Call
Control (DMCC). This platform provides complete backwards compatibility for all
these APIs ensuring that the Application Enablement Services platform will serve
legacy, current and future application needs.
The functionality provided by Application Enablement Services includes:
Connectors
Core and Administration Services
Communication Services Software Development Kit
Connectors communicate with the communication servers and expose standardsbased APIs that include:
Call Control is needed to perform third party call control operations
Device control is needed to gain exclusive or shared control of softphoneenabled Communication Manager telephones or extensions so as to perform
telephone operations using button presses, feature access codes, lamp,
ringer and display updates
Media Access is needed to perform media processing such as play, record,
media streaming to speaker/microphone
Management Interface is needed to facilitate Move, Add and Change (MAC)
of stations
System Administration Services provide infrastructure and house keeping functions
for the application platform including:
Data stores
Process life cycle management
Administration
Logging and Alarming
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Communication Services provide higher level of abstraction based on the APIs.
They expose functionality using Web Services interfaces. They allow easy
integration with enterprise applications and aggregation of service to implement
compound operations like ClickToCommunicate. Communication Services include:
Telephony Service - enables high level call control functionality over standard
web services interfaces (SOAP/XML)
User Service - enables a single, shared, user identity concept for users of
Avaya communication services and applications and integration with Identity
Management systems
System Management Service - exposes management features of Avaya
Communication Manager
Application Enablement Services software development kits (SDKs) consist of the
client API libraries, XSDs, WSDL, programmer's guides, sample applications,
simulators and other development tools. The following three SDKs are available for
Application Enablement Services 3.0:
IP Communications SDK (Device, Media and Call Control)
TSAPI/JTAPI SDK
Web Services SDK
Both Contact Center Express and Avaya Interaction Center support screen pop and
data directed routing for the Avaya call center using the Avaya Application
Enablement Services.
Avaya Contact Center Express
The Contact Center Express Call Routing Server enables intelligent call routing for
inbound calls. The routing is based on received call data matched with customer
information, call center statistics or agent availability.
The Call Routing Server:
Manages the Avaya Computer Telephony connection
Monitors VDNs
Manages VDN licensing
Registers for routing services
Receives call events
Issues routing instructions
Loads (manages) generic extensions, such as the Generic SQL Extension,
which gives the server access to SQL Server databases.
The Call Routing Server supports a single connection to a switch/media server. This
connection can be backed up via a standby link.
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The Routing Server uses the Application Server Generic (ASG) SQL Extension to
access information stored in any SQL Server database. The database could be a
Contact Center Express database or any other type, for example, a Microsoft Access
database.
Using a routing link extension, a call vector sends an adjunct routing request to the
Telephony Server, which, in turn, informs the Routing Server that has registered
the associated VDN. The Routing Server handles the request (usually to a
database) for information relating to the collected digits.
The database passes back the extension number the call should be sent to (this
could be a skill, split, agent DN, DDI or international number). The number is then
processed by the Routing Server and a route selection request is passed to the
Telephony Server and onto the switch/media server.
If the switch/media server doesn't receive the call destination within the wait-time
specified in the vector, it processes the next command in the list.
XML Server – The New Standard for Information Exchange
The eXtensible Markup Language (XML) has quickly become the standard for
information exchange between disparate devices. This mechanism has been chosen
by the European Computer Manufacturers Association (ECMA) as a standard for
interfacing computer telephony.
The XML Service consists of the XML Server, which converts the existing CSTA II
interface of Avaya Computer Telephony software to CSTA III XML, and XML Client,
which allows developers to build CTI applications in .Net.
This CSTA XML-over-TCP interface complies with ecma-269, ecma-285 and ecma323 (specifically as described in Annex G of the Standard ECMA - 323 June 2001,
XML Protocol for Computer Supported Telecommunications Applications (CSTA)
Phase III.)
XML Server is a Windows service that starts with the operating system. On startup,
it retrieves all configuration data from its local configuration file. Each XML client
that connects to the XML Server opens a corresponding link to a Telephony Server.
This connection opens using a single user name and password provided in the
configuration data. The supplied user name/password combination enables access
to all appropriate Avaya devices via the security database. XML Server supports
connections to multiple Telephony Servers.
Distributed as part of the Developer toolkit, XML Client provides a CSTA III XML
interface that allows developers to build Windows-based CTI applications in
Microsoft Visual .Net or Visual C#.
XML Client encompasses four developer components
XML Client is the core component that communicates with the XML Server.
Representing the base-level XML/CSTA tier, developers can use XML Client's
exposed objects to implement telephony operations directly, or they can
treat it as a 'data source' when using the higher-level, device-tier
components: XML Station, XML Routing and XML VDN.
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XML Station binds with XML Client to perform telephony operations on a
voice station and manage the calls associated with it. The objects exposed by
XML Station preserve the active calls on the voice station and allow users to
manipulate calls through a set of methods at the call appearance level.
XML Routing binds with XML Client to perform the telephony operations on a
registered VDN and manage the routing of calls associated with it.
XML VDN binds with XML Client to monitor VDNs (vector directory numbers)
and receive call events associated those VDNs.
SQL Plug-in – Integrating Relational Databases
The SQL Plug-in is a simple plug-in mechanism that allows you to integrate Avaya
Contact Center Express server applications with any SQL Server database without
the need for new development on the server.
This plug-in can be plugged in to any Contact Center Express server application that
supports the Plug-in Manager, such as IVR Server and the Call Routing Server.
SQL Plug-in uses Microsoft ADO to connect to a database and allow simple SQL
functionality to be available to the controlling application. The plug-in's detailed
configuration set allows named events to be received from the controlling
application. The events and associated parameters are converted to a direct SQL
statement which is then passed to the database for processing. Returned results
are extracted from the returned record set and passed back to the controlling
application via an associated return event.
Agent Rules and Rules Wizard
“How do I do a screen pop?” The Agent Rules capability is designed to allow you to
create a simple set of treatments for calls that meet fixed criteria. You can
configure, enable, disable and remove these rules using a simple GUI. Agent Rules
is similar to the email rules capability found in the Microsoft Outlook product.
The Agent Rules Wizard provides a configuration interface that allows rules to be
effectively managed. The Active Agent Rules Wizard steps the user through building
a valid rule by using a simple wizard mechanism. Simple administration allows the
user to add, enable, disable, edit and remove rules. A user can have multiple rules
that will be ordered based on entry.
Essentially, a rule fits into a simple statement; when a certain event occurs and a
call property matches this value, do this action then either continue rules
processing, jump to another rule or stop. You can create multiple rules for each
call event. The Agent Rules Plug-in processes them in the order in which they
appear in the Rules Wizard interface. Once a match is found, that action is taken
and no further rules are processed. You can change the processing order at any
time.
For example, a customer may configure a rule to deflect calls from a specific caller
number to voicemail. This rule would read:
When CallAlerting and CallerDN = 12345 do ReturnEvent Deflect %VoiceMail%
December 1, 2006
©2006 Avaya Inc.
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Request for Proposal for an IP Telephony System
Part 1 – Section 8 - Contact Center
When the CallAlerting event triggers this rule, the CallerDN property will be checked
to see if it matches “12345”. If it does then this call will be deflected to the value
contained in the VoiceMail configuration parameter.
The Agent Rules Wizard
Avaya Interaction Center
The Interaction Center desktop management application, Avaya Agent, actively
manages the agent’s desktop. Avaya Agent comes with the EDU viewer, an Avaya
Softphone, outbound list management (if a softdialer is enabled), and a prompter
that provides agent-prompting functionality. Custom applications can be integrated
with Avaya Agent and can help to cross-populate data between applications. Avaya
Agent can also be incorporated with Avaya Email Management and Avaya Web
Management, providing agents with a means of responding to emails and Web
queries all via the same desktop interface.
December 1, 2006
©2006 Avaya Inc.
Page 357
VoiceCon
Request for Proposal for an IP Telephony System
Part 1 – Section 8 - Contact Center
The Avaya Interaction Center solution supports screen pops with simultaneous
contact arrival at the agent desktop. Avaya Agent controls the agent desktop and
provides all the information necessary for agents to offer customers personalized
service while optimizing up-selling opportunities. Through screen pop, Avaya Agent
displays customer information, phone controls, and interaction scripts. Additionally,
Avaya Agent provides interfaces to back-office systems, allowing your agents to
easily and simultaneously display multiple windows showing customer information,
legacy data, and third-party applications.
The Avaya Interaction Center solution unifies the call center’s telephony and data
environments by creating the EDU (Electronic Data Unit) for each and every call.
The electronic data unit can be used to store call information about the caller’s IVR
activity, agent actions, and data mined from a variety of databases and platforms.
That data can then be retrieved from the electronic data unit and displayed on the
agent’s screen, placed in a database, inserted in a report, or used as search keys to
automatically access and screen pop appropriate responses from a database,
document, or application. The agent’s electronic data unit-based screen pop would
include all available account information tied to the caller’s subject, equipment, and
customers. The Avaya Softphone engine can also communicate with agent desktop
applications and automatically initiate the retrieval of the caller’s live account
screen(s) to the agent’s desktop.
Avaya Agent turns the agent’s screen into a Windows-like display showing different
types of information in separate sections. Customer data—such as customer name
and account number—is displayed in a frame that appears on one side of the screen
and remains in view during the entire customer contact session. The middle portion
of the screen can perform a number of tasks, such as displaying a Web chat in
progress, showing a complete customer history from your company’s database, or
popping up information from a knowledge repository. The bottom of the screen
displays Interaction Center’s powerful prompting tool, which provides agents with
customized scripts or prompts them to ask customers for information. The Avaya
Agent screen changes dynamically, depending on the media channel currently in
use. So your agents see only the information that’s appropriate—a key feature that
helps agents multitask effectively, without being overloaded with irrelevant data.
December 1, 2006
©2006 Avaya Inc.
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Request for Proposal for an IP Telephony System
Part 1 – Section 8 - Contact Center
Agents are “popped” the appropriate
application interfaces based on
workflow definitions and/or transaction
type. Appli cations can be pop ulated
automatically reducing redundant
keyboard entry.
Agents are intelligently “queued”
contacts th at match th eir skill-s ets and
are presented i nformation about each
contact (phone, email, web).
BMCC
Tools speci fic to the type of co nt act
are presented to assist the agent in
process ing t he contact efficiently.
Console Tools i nclude a “soft
tel ephone,” email response manager
and web t ext chat manager.
Agents are presented information to:
answer questions
guide t hem through transactions
present cross-sell messages
This helps reduce training requirements and
provides consistency in procedures.
Agents are presented a summary di splay of
contact/transaction history (including web, email &
pho ne contacts) enabling a complete view of the
customer’s interaction history. Ag ents have “oneclick” access t o int eraction details.
December 1, 2006
Agents are pres en ted with the current contact details
along wi th key information collected from customer
information systems. The res ult is a complete view of
the customer and contact providi ng agents t he
information needed to effecti vely service the customer.
©2006 Avaya Inc.
Page 359
VoiceCon
Request for Proposal for an IP Telephony System
Part 2 - System Pricing
Part 2:
1.0
System Pricing
System Pricing Requirements
Summary system and voice terminal pricing data will be presented to VoiceCon workshop
attendees and be deemed for public use.
Detailed pricing data will remain confidential, and used to verify if the proposed system
configurations satisfy RFP requirements.
Installation fee pricing data is required, and must be included in the RFP response. Indicate if the
proposed installation fee is based on direct sales/service or a channel partner pricing schedule.
The proposed system price must also include a 1-year warranty to the customer. If this is
a pricing option in your pricing schedule include it as part of the installation fee, and
identify it as such.
Avaya Response:
Comply; this is a direct sale through Avaya.
Avaya will provide Media or Parts replacement as well as access to software or
firmware downloads and access to Avaya’s Self Help Web site for the hardware
solution and CM application software all have a 1 year warranty. It is strongly
recommended that our customer purchase a Support Agreement coincident with
product purchase.
2.0
Summary Pricing – VOICECON NETWORK (all five locations)
Complete 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
Offer Price
All Common Equipment
(call processing, port interfaces, media gateways,
housings, power, feature/application servers, et al)
Generic Software (Standard Features)
Optional Software (Including License Fees)
Desktop Voice Terminals
Systems Management/Administration System
Messaging System
Installation Fee (including 1-year warranty)
Avaya Response:
Comply; please see Avaya_VoiceCon07_Pt 02 System Pricing.xls file for requested
Information.
December 1, 2006
©2006 Avaya Inc.
Page 360
VoiceCon
Request for Proposal for an IP Telephony System
Part 2 - System Pricing
3.0
Desktop Voice Terminal Pricing
Complete 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
Economy Desktop IP Telephone Instrument
Administrative Desktop IP Telephone Instrument
Professional Desktop IP Telephone Instrument
Executive Desktop IP Telephone Instrument
IP Audio conferencing Unit
PC Client Softphone (Station User) License Fee
PC Client Softphone (Attendant) License Fee
Key Module Add-on
Gigabit Ethernet Module Add-on
Display Module Add-on
WLAN Module Add-on
Desktop Power Module Option
Avaya Response:
Comply; please see Avaya_VoiceCon07_Pt 02 System Pricing.xls file for requested
Information.
December 1, 2006
©2006 Avaya Inc.
Page 361
VoiceCon
Request for Proposal for an IP Telephony System
Part 2 - System Pricing
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 to verify adherance to system
configuration performance requirements and the pricing summary data.
Avaya Response:
Comply; please see Avaya_CONFIDENTIAL_Pricing_VoiceCon
requested Information.
December 1, 2006
©2006 Avaya Inc.
RFP.xls
file
for
Page 362
Section 2.0
System Summary Pricing
LIST
DISCOUNTED
NOTES
All Common Equipment
(call processing, port interfaces, media gateways,
housings, power, feature/application servers, et al)
Generic Software (Standard Features)
Optional Software Features/Packages
IP Port License Fees (if applicable)
Desktop Voice Terminals
$337,315.00
$212,508.45
$51,600.00
$32,508.00
N/A
$
N/A
265,980.00
$167,567.00
$752,441.00
$474,037.83
Systems Management/Administration System
$0.00
$0.00
Messaging System
$181,033.17
$114,050.90
Installation Fee (including 1-year warranty)
$115,114.02
$97,846.92
$1,703,483.19
$1,073,194.41
TOTAL
Avaya Confidential
December 1, 2006
Universal
License for
all Ports
Entitlement
Zero Cost
Page 1 of 2
Section 3.0
Voice Terminals
LIST
DISCOUNTED
Economy Desktop IP Telephone Instrument
$139.00
$87.57
Administrative Desktop IP Telephone Instrument
$525.00
$330.75
Professional Desktop IP Telephone Instrument
$625.00
$393.75
Executive Desktop IP Telephone Instrument
$679.00
$427.77
$1,299.00
$818.37
$130.00
$81.90
$2,095.00
$1,819.85
Key Module Add-on
$225.00
$141.75
Gigabit Ethernet Module Add-on (if available)
$150.00
$94.50
Display Module Add-on (if available)
N/A
N/A
WLAN Module Add-on (if available)
N/A
N/A
$29.00
$93.00
$18.27
$58.59
IP Audioconferencing Unit
PC Client Softphone (Station User) License Fee
PC Client Softphone (Attendant) License Fee
Desktop Power Module Option (if available)
Basic
With Battery Backup
Avaya Confidential
December 1, 2006
Page 2 of 2
Avaya Communication Manager Feature Overview
Application Programming Interface Features
Avaya Communication Manager Feature Overview
Application Programming Interface
An application programming interface (API) allows numerous software applications to work with Avaya Communication
Manager. APIs also allow a client programmer to create their own applications that work with Communication Manager.
Application Enablement Application Enablement Services (AE Services) is a connector that provides connectivity
Services
between applications and Communication Manager. This connector allows development of
new applications and new features without having to modify Communication Manager or
expose its proprietary interfaces.
AE Services provides a single common platform architecture for call control, device control,
media control, and management. AE Services enables internal Avaya developers and
external partners to create powerful applications that harness the extensive Communication
Manager feature set.
Avaya offers software-only or bundled server AE Services deployment options. The same
client applications and software development kits (SDK) can run against both options.
CVLAN
CallVisor LAN (CVLAN) is an application programming interface (API) that enables
applications to communicate with Communication Manager. CVLAN sends and receives ASAI
messages over shared ASAI links on TCP/IP. An application can use ASAI messages to
monitor and control Communication Manager resources.
CVLAN software consists of a client component and a server component. The CVLAN client
can be installed on a server or on a client workstation. The CVLAN client provides clients with
access to the switch using the CVLAN server.
Web services
Telephony Service
Telephony Service (TS) is a web service that exposes basic outbound call control features of
Communication Manager. Telephony Service enables its clients to originate an outbound call,
drop a call, transfer a call, or conference a party into a call.
Telephony Service is one of the web services that resides on the Application Enablement
Services platform (AE Services).
System Management
System Management Service (SMS) is a web service that exposes management features of
Service
Communication Manager to clients. SMS enables its clients to display, list, add, change, and
remove specific managed objects on Communication Manager that are available through the
OSSI protocol and SAT screens.
SMS is one of the web services that resides on the Application Enablement Services platform
(AE Services).
User Service
User Service provides a common way of administering, retrieving, and programmatically
operating on user data. User Service provides a common user store and a programmable
interface for products and applications with which to integrate. User Service has a common
industry-standard data store (LDAP) as the repository for common user profile data.
User Service has web services as the infrastructure. This infrastructure allows products to
integrate with User Service at your schedule. User Service exposes a programmatic SOAP
interface that allows clients to write third party applications to utilize its functionality.
This integration occurs through the use of software adapters to User Service. The adapter
and web services technology allows User Service to publish user events to the product
spaces, and the product spaces to publish events to the common user area.
So if an administrator adds a user to the common store, a user event is sent to all
participating products with the appropriate information. Likewise, if a product level
administrator modifies a user record in its own user system, an event is sent to User Service
for the modified data to be stored in the common store. User Service then relays this user
event to the other participating product areas.
©2006 Avaya Inc.
Page 1
Avaya Communication Manager Feature Overview
Application Programming Interface Features
Application Programming Interface
An application programming interface (API) allows numerous software applications to work with Avaya Communication
Manager. APIs also allow a client programmer to create their own applications that work with Communication Manager.
Device and media
Device and media control API provides a connector to Communication Manager that allows
control API
clients to develop applications that provide first party call control. Applications can register as
IP extensions on Communication Manager and then monitor and control those extensions.
Device and media control API consists of connector server software and a connector client
API library. The connector server software runs on a hardware server that is independent
from Communication Manager. That is, device and media control API does not run coresident with Communication Manager.
DEFINITY LAN Gateway DEFINITY LAN Gateway (DLG) is a software service that tunnels the ASAI call control
protocol messages onto IP packets for transport between a customer Computer Telephony
Integration (CTI) server or application and Communication Manager.
Adjunct switch
Adjunct Switch Application Interface (ASAI) links Communication Manager and adjunct
application interface
applications. The interface allows adjunct applications to access Communication Manager
features and supply routing information to the system.
ASAI is the Avaya recommendation for Computer Telephony Integration (CTI). ASAI is based
on the Q.932 protocol.
JTAPI
Java telephony application programming interface (JTAPI) is an open API supported by
Avaya computer telephony that enables integration to Communication Manager ASAI. It is an
object-oriented programming interfaces favored for the development of multimedia solutions.
JTAPI applications are supported on any clients that supports a JAVA virtual machine (this
includes Windows, UnixWare, and Solaris platforms), or a Java-compatible Web browser.
TSAPI
Telephony Services Application Programming Interface (TSAPI) is an open API supported by
Avaya computer telephony that allows integration to Communication Manager ASAI.
TSAPI is based on international standards for CTI telephony services. Specifically, the
European Computer Manufacturers Association (ECMA) CTI standard definition of ComputerSupported Telecommunications Applications (CSTA) is the foundation for TSAPI. The CSTA
standard is a technical agreement reached by an open, multi-vendor consortium of major
switch and computer vendors. Since CSTA Services and protocol definitions are the basis for
TSAPI, TSAPI provides a generic, switch-independent API. CSTA services logically integrate
the two most common pieces of equipment on user desktops, the telephone and the personal
computer.
Security administration for telephony services allows administrators to restrict user access to
TSAPI features in various ways. For example, an administrator might restrict a user to control
and monitoring of the telephone at their desktop. Similarly, an administrator can restrict a user
to call control and monitoring of the telephone at any desktop where they log in.
Expanded security permissions can increase user control in support of work group or
departmental telephony applications. Administrators can expand user permissions even
further to include any telephone or device that it is possible to control on a CTI link. An
administrator might assign an unrestricted security permission level to a server application
that processes calls before call delivery to user desktops in a call center environment. An
administrator can assign different users different permissions.
©2006 Avaya Inc.
Page 2
Avaya Communication Manager Feature Overview
Attendant Features
Attendant Features
Avaya Communication Manager contains many exciting features that provide easy ways to communicate through your
telephone system attendant (operator). In addition, attendants can connect to their console (switchboard) from other
telephones in your system, thereby expanding the attendant capabilities.
Accessing the Attendant
Dial access to
The dial access to attendant feature allows you to reach an attendant by dialing an access
attendant
code. The attendant can then extend the call to a trunk or to another telephone.
Individual attendant
Individual attendant access allows you to call a specific attendant console. Each attendant
access
console can be assigned an individual extension number.
Recall
This feature allows users to recall the attendant when they are on a two-party call or on an
attendant conference call held on the console.
•
Single-line users press the recall button or flash the switch hook to recall the
attendant.
•
Multi-appearance users press the conference or transfer button to recall the
attendant and remain on the connection when either button is used.
Attendant backup
The attendant backup feature allows you to access most attendant console features from one
or more specially-administered backup telephones. This allows you to answer calls more
promptly, thus providing better service to your guests and prospective clients.
When the attendant console is busy, you can answer overflow calls from the backup
telephones by pressing a button or dialing a feature access code. You can then process the
calls as if you are at the attendant console. The recommended backup telephones are the
Avaya models 6408, 6416, or 6424.
Attendant room status
Communication Manager allows an attendant to see whether a room is vacant or occupied,
and what the housekeeping status of each room is. This feature is available only when you
have enhanced hospitality enabled for your system.
This feature combines the property management capabilities of housekeeping status and
check-in/check-out, but does not require that you have a property management system
(PMS).
Attendant functions using Distributed Communications System protocol
Control of trunk group
Control of trunk group access allows an attendant at any node in the Distributed
access
Communications System (DCS) to take control of any outgoing trunk group at an adjacent
node. This is helpful when an attendant wants to prevent telephone users from calling out on
a specific trunk group for any number of reasons, such as reserving a trunk group for
incoming calls or for a very important outgoing call.
Direct trunk group
Direct trunk group selection allows the attendant direct access to an idle outgoing trunk in a
selection
local or remote trunk group by pressing the button assigned to that trunk group. This feature
eliminates the need for the attendant to memorize, or look up, and dial the trunk access codes
associated with frequently used trunk groups. Direct trunk group selection is intended to
expedite the handling of an outgoing call by the attendant.
Inter-PBX attendant
Inter-PBX attendant calls allows attendants for multiple branches to be concentrated at a main
calls
location. Incoming trunk calls to the branch, as well as attendant-seeking voice-terminal calls,
route over tie trunks to the main location.
Call handling
Attendant Intrusion
Use the Attendant Intrusion feature to allow an attendant to intrude on an existing call. The
Attendant Intrusion feature is also called Call Offer.
Attendant lockout This feature prevents an attendant from re-entering a multiple-party connection held on the
privacy
console unless recalled by a telephone user. This feature is administered on a system-wide
basis. It is either activated or not activated.
©2006 Avaya Inc.
Page 3
Avaya Communication Manager Feature Overview
Attendant Features
Attendant Features
Avaya Communication Manager contains many exciting features that provide easy ways to communicate through your
telephone system attendant (operator). In addition, attendants can connect to their console (switchboard) from other
telephones in your system, thereby expanding the attendant capabilities.
Attendant split swap
The attendant split swap feature allows the attendant to alternate between active and split
calls. This operation may be useful if the attendant needs to transfer a call but first must talk
independently with each party before completing the transfer.
Attendant vectoring
Attendant vectoring provides a highly flexible approach for managing incoming calls to an
attendant. For example, with current night service operation, calls redirected from the
attendant console to a night station can ring only at that station and will not follow any
coverage path.
With attendant vectoring, night service calls will follow the coverage path of the night station.
The coverage path could go to another station and eventually to a voice mail system. The
caller can then leave a message that can be retrieved and acted upon.
Automated attendant
Automated attendant allows the calling party to enter the number of any extension on the
system. The call is then routed to the extension. This allows you to reduce cost by reducing
the need for live attendants.
Backup alerting
The backup alerting feature notifies backup attendants that the primary attendant cannot pick
up a call. It provides both audible and visual alerting to backup stations when the attendant
queue reaches its queue warning level. When the queue drops below the queue warning
level, alerting stops.
Audible alerting also occurs when the attendant console is in night mode, regardless of the
attendant queue size.
Call waiting
Call waiting allows an attendant to let a single-line telephone user who is on the telephone
know that a call is waiting. The attendant is then free to answer other calls. The attendant
hears a call waiting ringback tone and the busy telephone user hears a call waiting tone. This
tone is heard only by the called telephone user.
Calling of inward
A telephone with a class of restriction (COR) that is inward restricted cannot receive public
restricted stations
network, attendant-originated, or attendant-extended calls. This feature allows you to override
this restriction.
Conference
The conference feature allows an attendant to set up a conference call for as many as six
conferees, including the attendant. Conferences from inside and outside the system can be
added to the conference call.
Starting with Communication Manager release 3.0, attendants can set up conferences for
more than six people using the Enhanced Meet-me Conferencing feature.
Listed directory
Allows outside callers to access your attendant group in two ways, depending on the type of
number
trunk used for the incoming call. You can allow attendant group access through incoming
direct inward dial trunks, or you can allow attendant group access through incoming central
office and foreign exchange trunks.
Override of diversion
The override of diversion feature allows an attendant to bypass diversion features such as
features
send all calls and call coverage by putting a call through to an extension even when these
diversion features are on. This feature, together with attendant intrusion, can be used to get
an emergency or urgent call through to a telephone user.
Priority queue
Priority queue places incoming calls to the attendant in an orderly queue when these calls
cannot go immediately to the attendant. This feature allows you to define twelve different
categories of incoming attendant calls, including emergency calls, which are given the highest
priority.
©2006 Avaya Inc.
Page 4
Avaya Communication Manager Feature Overview
Attendant Features
Attendant Features
Avaya Communication Manager contains many exciting features that provide easy ways to communicate through your
telephone system attendant (operator). In addition, attendants can connect to their console (switchboard) from other
telephones in your system, thereby expanding the attendant capabilities.
Release loop
Release loop operation allows the attendant to hold a call at the console if the call cannot
operation
immediately go through to the person being called. A timed reminder begins once the call is
on hold. If the call is not answered within the allotted time, the call returns to the queue for the
attendant. Timed reminders attempt to return the call to the attendant who previously handled
it. Only when the original attendant is unavailable are calls returned to the queue.
Selective conference
See Selective conference mute.
mute
Serial calling
The serial calling feature enables an attendant to transfer trunk calls that return to the same
attendant after the called party hangs up. The returned call can then transfer to another
station within the switch. This feature is useful if trunks are scarce and direct inward dialing
services are unavailable. An outside caller may have to redial often to get through because
trunks are so busy. Once callers get through to an attendant they can use the same line into
the switch for multiple calls. The attendant display shows if an incoming call is a serial call.
Timed reminder and
Attendant timers automatically alert the attendant after an administered time interval for the
attendant timers
following types of calls:
Centralized Attendant
Service
Display
Auto Start and Do Not
Split
Auto Manual Splitting
•
Extended calls to be answered or waiting to be connected to a busy single-line
telephone
•
One-party calls placed on hold on the console
• Transferred calls that have not been answered after transfer
The timed reminder feature informs the attendant that a call requires additional attention. After
the attendant reconnects to the call, the user can either choose to try another extension
number, hang up, or continue to wait. Communication Manager supports a variety of
administrable attendant timers for use in a variety of situations.
Centralized Attendant Service (CAS) enables attendant services in a private network to be
concentrated at a central location. Each branch in a centralized attendant service has its own
listed directory number or other type of access from the public network. Incoming calls to the
branch, as well as calls made by users directly to the attendants, are routed to the centralized
attendants over release link trunks.
The display feature shows call-related information that helps the attendant to operate the
console. This feature also shows personal service and message information. Information is
shown on the alphanumeric display on the attendant console. Attendants may select one of
several available display message languages: English, French, Italian, or Spanish. In addition,
your company may define one additional language for use by users and attendants on their
display.
Making calls
The Auto Start feature allows the attendant to make a telephone call without pushing the start
button first. If the attendant is on an active call and presses digits on the keypad, the system
automatically splits the call and begins dialing the second call.
The Do Not Split feature deactivates the auto start feature and allows the sending of touch
tones over the line for the purposes of such things as picking up messages.
Auto Manual Splitting allows an attendant to announce a call or consult privately with the
called party without being heard by the calling party on the call. It splits the calling party away
so the attendant can confidentially determine if the called party can accept the call.
©2006 Avaya Inc.
Page 5
Avaya Communication Manager Feature Overview
Attendant Features
Attendant Features
Avaya Communication Manager contains many exciting features that provide easy ways to communicate through your
telephone system attendant (operator). In addition, attendants can connect to their console (switchboard) from other
telephones in your system, thereby expanding the attendant capabilities.
Monitoring calls
Attendant control of
Use the Attendant Control of Trunk Group Access feature to allow the attendant to control
trunk group access
outgoing and two-way trunk groups. The attendant usually activates this feature during
periods of high use. This is helpful when an attendant wants to prevent telephone users from
calling out on a specific trunk group. Some reasons are to reserve a trunk group for incoming
calls or for a very important outgoing call.
This feature also prevents telephone users from directly accessing an outgoing trunk group
that the attendant has controlled.
Attendant direct
This feature allows the attendant to keep track of extension status -whether the extension is
extension selection
idle or busy -and to place or extend calls to extension numbers without having to dial the
extension number. The attendant can use this feature in two ways:
•
Attendant direct trunk
group selection
Crisis alerts to an
attendant console
Trunk group
busy/warning
indicators to attendant
Trunk identification by
attendant
using standard direct extension selection access
• using enhanced direct extension selection access
With this feature, the attendant directs access to an idle outgoing trunk by pressing the button
assigned to the trunk group. This feature eliminates the need for the attendant to memorize,
or look up, and dial the trunk access codes associated with frequently used trunk groups.
Pressing a labeled button selects an idle trunk in the desired group.
Crisis alert uses both audible and visual alerting to notify attendant consoles when an
emergency call is made. Audible alerting sounds like an ambulance siren. Visual alerting
flashes the CRSS-ALRT button lamp and the display of the caller name and extension (or
room). The display of the origin of the emergency call enables the attendant or other user to
direct emergency service response to the caller. Though often used in the hospitality industry,
it can be set up to work with any standard attendant console.
When crisis alerting is active, the console is placed in position-busy mode so that other
incoming calls can not interfere with the emergency call notification. The console can still
originate calls to allow notification of other personnel. Once a crisis alert call has arrived at a
console, the console user must press the position-busy button to un-busy the console, and
press the crisis-alert button to deactivate audible and visual alerting.
If an emergency call is made while another crisis alert is still active, the incoming call will be
placed in the queue. If the system is administered so that all users must respond, then every
user must respond to every call, in which case the calls are not necessarily queued in the
order in which they were made. If the system is administered so that only one user must
respond, the first crisis alert remains active at the telephone where it was acknowledged.
Subsequent calls are queued to the next available station in the order in which they were
made.
This feature provides the attendant with a visual indication that the number of busy trunks in a
group has reached an administered level. A visual indication is also provided when all trunks
in a group are busy. This feature is particularly helpful to show the attendant that the
attendant control of trunk group access feature needs to be invoked.
Trunk identification allows an attendant or display-equipped telephone user to identify a
specific trunk being used on a call. This capability is provided by assigning a trunk ID button
to the attendant console or telephone. This feature is particularly helpful for identifying a faulty
trunk. That trunk can then be removed from service and the problem quickly corrected.
©2006 Avaya Inc.
Page 6
Avaya Communication Manager Feature Overview
Attendant Features
Attendant Features
Avaya Communication Manager contains many exciting features that provide easy ways to communicate through your
telephone system attendant (operator). In addition, attendants can connect to their console (switchboard) from other
telephones in your system, thereby expanding the attendant capabilities.
Visually Impaired
Visually Impaired Attendant Service (VIAS) provides voice feedback to a visually impaired
Attendant Service
attendant. Each voice phrase is a sequence of one or more single-voiced messages. This
feature defines six attendant buttons to aid visually impaired attendants:
•
Visually impaired service activation/deactivation button: activates or deactivates the
feature. All ringers previously disabled (for example, recall and incoming calls)
become re-enabled.
•
Console status button: voices whether the console is in position available or position
busy state, whether the console is a night console, what the status of the attendant
queue is, and what the status of system alarms is.
•
Display status button: voices what is shown on the console display. VIAS support is
not available for all display features (for example, class of restriction information,
personal names, and some call purposes).
•
Last operation button: voices the last operation performed.
•
Last voiced message button: repeats the last voiced message.
•
New reason code for
attendant vector
Direct trunk group selection status button: voices the status of an attendantmonitored trunk group.
The visually impaired attendant may use the Inspect mode to locate each button and
determine the feature assigned to each without actually executing the feature.
Do Not Disturb (DND) calls that are routed to an attendant vector now display the reason
code ct. This reason code is consistent with the display for an attendant console.
©2006 Avaya Inc.
Page 7
Avaya Communication Manager Feature Overview
Call Center Features
Call Center
The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of
features, capabilities, and applications designed to meet all of your customers’ call center needs.
Computer Telephony
Computer Telephony Integration (CTI) enables Communication Manager features to be
Integration
controlled by external applications, and allows integration of customer databases of
information with call control features.
Avaya Computer Telephony (formally named CentreVu™ Computer Telephony) is server
software that integrates the premium call control features of Communication Manager with
customer information in customer's databases. It is a local area network (LAN)-based CTI
solution consisting of server software that runs in a client/server configuration. Avaya
Computer Telephony delivers the CTI architecture and platform that supports contact center
application requirements, along with emerging applications programming interfaces (APIs).
Adjunct route support
This feature provides the capability to invoke Network Call Redirection (NCR) through the
for network call
route request response to an adjunct route vector step. This allows a CTI application to
redirection
directly utilize NCR for redirecting an incoming call in the PSTN through the ASAI adjunct
routing application.
The redirection request, along with the PSTN redirected to a telephone number, is included in
the route select message from the adjunct. The redirect request invokes whatever form of
network redirection that is assigned to the trunk group for the incoming call in the same
manner as a vector invoked NCR. Information forwarding to the redirected destination is
supported in the same manner as a vector invoked NCR.
This capability functions with either the network transfer type where the switch sets up the 2nd
leg of a call, or the network deflection type where the PSTN sets up the 2nd leg of a call of
NCR protocols.
ASAI support for Aux
You can now assign up to 99 Aux Work reason codes, rather than only 10. The description for
Work reason codes
each reason code can now be up to 16 characters, rather than only 10 characters.
Note: ASAI does not currently support two-digit reason codes.
To take advantage of the additional reason codes, set the:
•
Reason Codes field on the Customer Options screen to y
•
Block CMS Move Agent
events
Expert Agent Selection (EAS) Enabled and the Two-Digit Aux Work Reason Codes
fields on the Feature-Related System Parameters screen to y.
This feature lets you prevent the system from sending the ASAI logout-login event messages,
that are related to an agent move. When this CTI link option is activated, the changes to the
agent state, such as logout followed by login and return to previous state, will not be reported
to the ASAI adjunct. This operation is required by Avaya IC since the initial logout causes IC
to permanently logout the agent, disrupting normal operation. IC does not need to be informed
of agent skill moves via this method. This option will be available to other applications for use
where needed.
©2006 Avaya Inc.
Page 8
Avaya Communication Manager Feature Overview
Call Center Features
Call Center
The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of
features, capabilities, and applications designed to meet all of your customers’ call center needs.
Co-resident DEFINITY
In simplest terms, the DEFINITY Local Area Network (LAN) Gateway, or DLG, is an
LAN Gateway
application that enables communications between TCP/IP clients and Communication
Manager call processing. In more technical terms, the DLG application is software that both
routes inter-network messages from one protocol to another (ISDN to TCP/IP) and bridges all
ASAI message traffic by way of a TCP/IP tunnel protocol.
In previous configurations, a DEFINITY LAN gateway (DLG) was connected externally on a
separate TN801 MAPD circuit pack. The DLG application is packaged internally where it coresides with the Communication Manager. The internally packaged DLG is referred to as the
co-resident DLG.
Co-resident DLG is only available with the S8300 Media Server.
Co-resident DLG provides the functionality of the Adjunct/Switch Application Interface (ASAI)
using an Ethernet transport instead of a Basic Rate Interface (BRI) transport. In the S8300
Media Server, connectivity is provided through the processor Ethernet.
Direct Agent
Announcement
Flexible billing
Pending work mode
change
Trunk group
identification
User-to-User
Information
propagation during
manual
transfer/conference
operations
VDN override for ASAI
messages
Direct Agent Announcement (DAA) enhances direct agent calling capabilities for Adjunct
Switch Application Interface (ASAI) and Expert Agent Selection (EAS). It plays an
announcement to direct agent callers waiting in a queue.
The flexible billing feature allows Communication Manager or an adjunct to communicate with
the public network using ISDN PRI messages to change the billing rate for an incoming 900type call. Rate-change requests to specify a new billing rate can be made anytime after a call
is answered and before it disconnects.
Flexible billing is available in the U.S. for use with AT&T MultiQuest 900 Vari-A-Bill service.
Flexible billing requires an adjunct switch application interface and other application software.
This feature allows ASAI applications to change the current work mode of an agent while that
agent is busy on a call. The change is a pending change that will take effect as soon as all the
current calls are cleared.
Trunk group identification provides ASAI applications with the capability to obtain trunk group
information even when the Calling Party Number (CPN) is known. ASAI will provide the trunk
group information in the event reports for both inbound and outbound calls. If the Automatic
Number Identification (ANI) is known, the event reports will contain the trunk group
information and the CPN.
This feature enables UUI, specifically used by ASAI, to be propagated to the new call during a
manual transfer or conference operation. Previously, ASAI UUI could not be sent in a setup
message when the call was transferred to another system, so the ASAI UUI was never
passed to an application monitoring calls on the system receiving the transfer.
This feature only applies to manual transfer and conference operations. If the transfer or
conference operation is controlled by a software application (for example, controlling calls or
agents over an ASAI link), the application can insert the desired ASAI UUI into the new call.
This feature provides a VDN option to override the called number in certain ASAI messages
for ISDN calls. This applies to CTI applications that require the active VDN extension instead
of the called number. This is a field of the VDN Screen - "VDN Override for ISDN Trunk ASAI
Messages. The default value is no.
For calls to VDNs with the option set to y(es), the called number provided will correspond to
the active VDN for call instead of the original called number provided in the incoming ISDN
SETUP message. This applies to the ASAI call-offered, alerting, queued and connect event
messages and the adjunct route-request message.
©2006 Avaya Inc.
Page 9
Avaya Communication Manager Feature Overview
Call Center Features
Call Center
The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of
features, capabilities, and applications designed to meet all of your customers’ call center needs.
Automatic Call
Automatic Call Distribution (ACD) is the basic building block for call center applications. ACD
Distribution
offers you a method for distributing incoming calls efficiently and equitably among available
agents. With ACD, incoming calls can be directed to the first idle or most idle agent within a
group of agents.
Agents in an ACD environment are assigned to a hunt group, a group of agents handling the
same types of calls. A hunt group is also known as a split or skill with Expert Agent Selection
(EAS).
Abandoned Call
Abandoned Call Search allows a central office that does not provide timely disconnect
Search
supervision to identify abandoned calls. An abandoned call is one in which the calling party
hangs up before the call is answered. Abandoned Call Search is suitable only for older central
offices that do not provide timely disconnect supervision.
Adjunct Routing
Adjunct Routing is a vector step that, when executed, sends a route request over the
specified link to the connected adjunct asking where to route the call being processed. The
adjunct is then to respond with a route-select message specifying the destination either
internal or outside number where the call is to be routed. Adjunct Routing is used in
conjunction with ASAI.
Auto-Available Split
Auto-Available Split (AAS) allows members of an Automatic Call Distribution (ACD) split to be
continuously in auto-in work mode. An agent in auto-in work mode becomes available for
another ACD call immediately after disconnecting from an ACD call. You can use AAS to
bring ACD-split members back into auto-in work mode after a system restart.
Although not restricted to such, this feature is intended to be used for splits containing only
recorders or voice-response units.
Automatic Number
Use the Automatic Number Identification (ANI) feature to display telephone number of the
Identification
calling party on your display telephone. The system uses ANI to interpret calling party
information that is signaled over multifrequency (MF) or Session Initiation Protocol (SIP)
trunks.
Incoming Automatic
Use in-band signaling for information, such as the address digits for the called party, that is
Number Identification
delivered over the same trunk circuit that is used for the voice or data connection. Use out-ofband or ISDN signaling when signaling information passes through a different signaling path
than the path that is used for the voice or data connection.
For example, when a call is made from 555-3800 to your display telephone at extension
81120, and the Incoming Tone (DTMF) ANI field is set to *ANI*DNIS* on the Trunk Group
screen, your trunk group receives *5553800*81120*. If the same field is set to ANI*DNIS*,
your trunk group receives 5553800*81120*. In both cases, call from 555-3800 appears on
your telephone display.
If you do not use in-band ANI, the incoming trunk group name appears on your telephone
display.
Outgoing Automatic
Outgoing ANI applies to outgoing Russian MF ANI, R2-MFC ANI, China #1 MF ANI, and
Number Identification
Spain Multi Frequency España (MFE) ANI trunks only.
Use Outgoing ANI to specify the type of ANI to send on outgoing calls. You can define MF
ANI (the calling party number, sent through multifrequency signaling trunks) prefixes by COR.
This allows a system to send different ANIs to different central offices (COs).
For a tandem call that uses different types of incoming and outgoing trunks, the server uses:
•
The COR-assigned call type of the incoming trunk for Russian or R2-MFC outgoing
trunks
•
Automatic Route Selection (ARS) call types for MFE outgoing trunks
©2006 Avaya Inc.
Page 10
Avaya Communication Manager Feature Overview
Call Center Features
Call Center
The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of
features, capabilities, and applications designed to meet all of your customers’ call center needs.
Local feedback for
A cost saving trend used by many call centers is the movement of agent seats from locations
queued ACD calls
in the US and EU to offshore locations. One detriment to achieving these savings is the
increase in trunk costs by redirecting calls to these offshore locations.
When a call is rerouted to an alternate switch, it becomes the responsibility of the destination
switch to provide audible feedback to the caller while that call remains in queue at the
destination switch waiting for an available agent. Typically, such audible feedback takes the
form of music interspersed with recorded announcements.
When the trunks between the sending and receiving switches are IP trunks, bandwidth is
utilized when the music and recorded announcement packets are sent from the destination
switch to the caller. Because of the continuous nature of music, the bandwidth required to
provide this audible feedback to callers in queue is generally greater than that required to
support a conversation between a caller and an agent.
Communication Manager allows vector processing to continue at the local sending switch,
even after a call has been routed to a queue on an offshore destination switch. Vector
processing at the sending switch can then continue to provide audible feedback to the caller
while the call is in queue at the destination switch. No packets need be sent over the IP trunk
during the queuing phase of the call.
Queue status
Communication Manager allows you to assign queue status indicators for ACD calls based on
indicators
the number of calls in queue and the time in queue. To help monitor queue activity, you can
assign these indications to lamps on agent, supervisor, or attendant terminals, or on consoles.
In addition, you can define auxiliary queue warning lamps to track queue status. On display
telephones, you can display the number of calls in queue, and the time in queue of the oldest
call.
Avaya Basic Call
The Avaya Basic Call Management System (BCMS) helps you fine tune your call center
Management System
operation by providing reports with the data necessary to measure your call center agents
performance.
The BCMS feature offers call management control and reporting at a low cost for call centers
of up to 2000 agents. BCMS collects and processes ACD call data (up to seven days) within
the system; an adjunct processor is not required to produce call management reports.
Avaya Business
Avaya Business Advocate is the collection of features that provide flexibility in the way a call
Advocate
is selected for an agent in a call surplus situation, and in the way an agent is selected for a
call. Instead of the traditional "first in, first out" approach, the needs of the caller, potential
business value, and the desire to wait are calculated. The system then decides what agents
should be matched to the callers.
Auto reserve agents
Auto reserve agents allows the system to use the percent allocation distribution feature for
agent skills.
Call selection override Call selection override is determined by skill. Call center supervisors can override the normal
per skill
call handling activity either on particular skills only, or for the entire call center.
Dynamic percentage
The dynamic percentage adjustment feature allows the system to compare actual levels of
adjustment
service with service targets. The system can then adjust the service target so that the overall
use of the skill is more efficient.
Dynamic queue
Dynamic queue position allows the system to put calls from multiple vector directory numbers
position
(VDNs) into a skill queue. The calculation is based on the ratio of ASA for the VDNs being
equal to the ratio of service objectives for the VDNs. This feature ensures balanced call
handling across VDNs.
©2006 Avaya Inc.
Page 11
Avaya Communication Manager Feature Overview
Call Center Features
Call Center
The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of
features, capabilities, and applications designed to meet all of your customers’ call center needs.
Dynamic threshold
Dynamic threshold adjustment allows the system to compare actual levels of service with
adjustment
service targets, and to adjust overload thresholds. This feature makes the use of overload
agents more efficient.
Logged-in advocate
The logged-in advocate agent counting feature counts agents toward the advocate agent limit
agent counting
if a service objective, percent allocation, or a reserved skill is assigned to the agent login ID,
or if one of the agent skills is assigned least occupied agent or service level supervisor.
Percent allocation
Percent allocation distribution allows the system to distribute calls to auto reserve agents by
distribution
comparing a reserve agent work time in a skill with the target allocation for that skill.
Reserve agent time in
This feature activates a reserve agent either if the expected wait time (EWT) exceeds a prequeue activation
determined threshold, or if the call time in the queue exceeds the administered service level
supervisor threshold. Reserve agents are then dropped off a skill only when both of the
following conditions are met:
•
The EWT for the skill drops below both administered thresholds.
•
Avaya call center
features supported on
the Avaya G700 Media
Gateway
Avaya Call
Management System
Avaya Virtual Routing
Enhanced information
forwarding
Call center release
control
The head call time in queue no longer exceeds the service level supervisor
threshold.
Avaya Call Center functionality is supported on the G700 Media Gateway with
Communication Manager, with either an S8300 Media Server or an S8700 Media Server.
The Avaya Call Management System (CMS) collects call traffic data, formats management
reports, and provides an administration interface for Automatic Call Distribution (ACD). It
helps you manage the people, traffic load, and equipment in an ACD environment by
answering such questions as:
•
How many calls are we handling?
•
How many callers abandon their calls before talking with an agent?
•
Are all agents handling a fair share of the calling load?
•
Are our lines busy often enough to warrant adding additional ones?
• How has traffic changed in a given ACD hunt group over the past year?
Avaya Virtual Routing (formerly known as Look-Ahead Interflow or LAI) balances the load of
ACD calls across multiple locations. Virtual routing helps customers balance call loads among
their locations by analyzing demand and directing each call to the location best able to handle
it -for example, based on call volume, waiting time in queue, or the time of day.
With Avaya virtual routing, you can optionally route a call to a backup location based on your
system ability to handle the call within parameters defined in a vector. In turn, the backup
system can accept or deny the call also based on defined parameters.
Avaya virtual routing allows interflowing of only the call(s) at or near the head of the queue to
provide First In/First Out (FIFO) call distribution and significantly reduce call and trunk
processing for Avaya virtual routing.
Enhanced information forwarding allows call center related information to be passed
transparently over some public networks and non-QSIG or QSIG private networks using
codeset 0 shared user-to-user information (UUI) (for non-QSIG) or QSIG manufacturerspecific information (MSI).
Call center release control determines which features are "active" on your switch. The call
center release control feature controls whether certain call center software features are
available to you.
©2006 Avaya Inc.
Page 12
Avaya Communication Manager Feature Overview
Call Center Features
Call Center
The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of
features, capabilities, and applications designed to meet all of your customers’ call center needs.
Call prompting
Call prompting allows the system to collect information from the calling party and direct the
calls using call vectoring.
The caller is verbally prompted by the system and enters information in response to the
prompts. This information is then used to redirect the call or handle the call in some other way
(taking a message, for example). This feature is mostly used to enhance the efficient handling
of calls in the automatic call distribution application.
Data collection
Data collection allows the calling party to enter data that can then be used by a host computer
application to assist in call handling. For example, this data may be the calling party account
number, which could then be used to support an inquiry/response application.
Data In/Voice Answer
Data In/Voice Answer (DIVA) allows the calling party to hear selected announcements based
on the digits that he or she enters. This may be used for applications such as an audio bulletin
board.
Call vectoring
Call vectoring is a versatile method of routing incoming calls that can be combined with
automatic call distribution for maximum benefit and call center efficiency. A call vector is a
series of call processing steps (such as providing ringing tones, busy tones, music,
announcements, and queuing the call to an ACD hunt group) that define how calls are
handled and routed. The steps, called vector commands, determine the type of processing
that specific calls will receive.
Vector commands may direct calls to on-premises or off-premises destinations, to any skill or
hunt group, or to a specific call treatment such as an announcement, forced disconnect,
forced busy, or music.
With combinations of different vector commands, incoming callers can be treated differently
depending on the time or day of the call, the expected wait time (EWT), the importance of the
call, or other criteria. Each vector can have up to 32 commands. Communication Manager
also allows vectors to be linked through the "goto vector" command.
Advanced vector
Advanced vector routing is a collection of features that enhance Communication Manager
routing
vector routing capabilities.
Average Speed of
Average Speed of Answer (ASA) routing is an enhancement to call vectoring that provides a
Answer routing
flexible method for routing calls or queuing calls based on their average speed of answer for a
VDN or a split/skill.
Best service routing
Best service routing (BSR) distributes the call to the best local or remote split/skill among the
resources to be considered, based on expected wait time (EWT) or available agent
characteristics.
Best service routing
Best service routing (BSR) polling over IP without B-channel provides the ability to do BSR
polling over IP without polling between multiple sites over H.323 IP trunks without requiring an ISDN PRI B-channel.
B-channel
This also eliminates the associated IP media processor hardware.
QSIG temporary signaling connections are used by the BSR polling software to eliminate the
need for the IP media processor board, thereby making BSR an even more cost effective
multi-site solution.
Expected Wait Time
The Expected Wait Time (EWT) feature makes call center routing decisions based on waiting
routing
time for calls in queue, using a patented algorithm that continuously estimates call waiting
times. Announcements of the wait time customers can expect before their call is answered
can make time in queue more tolerable.
Call center messaging Call center messaging gives the calling party the option of leaving a message or waiting in
queue for an agent. This may be used for an online order entry system or to further automate
an incoming call center operation.
©2006 Avaya Inc.
Page 13
Avaya Communication Manager Feature Overview
Call Center Features
Call Center
The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of
features, capabilities, and applications designed to meet all of your customers’ call center needs.
Holiday vectoring
With holiday vectoring, a flexible approach for managing incoming calls on special dates is
available. Holiday vectoring allows for branching and routing of calls based on information
about special schedules. The special schedules are recorded in tables, each of which can
hold up to 15 special dates or ranges of dates. Holiday vectoring makes it possible for up to
10 tables to be treated differently in vector processing.
Vector Directory
Calls access Communication Manager vectors using Vector Directory Numbers (VDN). A
Number
VDN is a "soft" extension number that is not assigned to a physical equipment location. A
VDN has several properties that are administered by the system manager.
A VDN can be accessed in almost any way that an extension can be accessed. When
answering a call, the answering agent sees the information (such as the name) associated
with the VDN on their display, and can respond to the call with knowledge of the dialed
number. This operation provides dialed number identification service (DNIS), allowing the
agent to identify the purpose of the incoming call.
Class of Restriction
Class of Restriction (COR) is checked for transfer to the VDN. It can also be used to block the
for VDN
AUX trunk announcement from some agents. Observing can also be set to allow or restrict to
that VDN.
Display VDN for route- Display VDN for route-to DAC provides a VDN option to have the display to the answering
to DAC
agent show the "caller to VDN" format. The option for the "caller to VDN" display is required
for ACD applications where a call needs to be routed to a specific agent, and have the call go
to coverage if the agent doesn’t answer or is logged out.
VDN in a coverage
VDN in a coverage path enhances call coverage and call vectoring to allow you to assign
path
vector directory numbers as the last point in coverage paths. Calls that go to coverage can be
processed by vectoring/prompting to extend call coverage treatments.
VDN of origin
VDN of origin announcement provides agents with a short message about the city of origin or
announcement
requested service or the caller, based on the VDN used to process the call. VOA messages
help agents respond appropriately to callers.
This feature is particularly useful for visually impaired agents or agents that do not have
display telephones.
VDN return
VDN return destination is an optional feature that re-routes a call that has been processed
destination
through a vector, to the administered return destination. This step occurs once all parties,
except the originator, have dropped. The return destination must be a VDN extension.
Call Work Codes
Call Work Codes (CWC) allows ACD agents to enter digits for an ACD call to record the
occurrence of a customer-defined event, such as a social security numbers or telephone
numbers. The agent enters the call work code by operating the CWC feature button and using
the dial pad during an ACD (inbound) call without interrupting the conversation, or in the After
Call Work (ACW) mode following the call. The digits are displayed on a display-equipped
telephone while being entered.
Caller Information
The Avaya call center also supports AT&T Caller Information Forwarding (CINFO) service,
Forwarding
allowing customers to collect customer-provided data forwarded through the network. This
information can be used to route calls or provide visual displays on agent voice terminals, or
be passed along to Computer Telephony Integration (CTI) applications.
Circular station hunt
This hunt group type is an alternative to the "ddc" or "hot-seat" algorithm in a hunt group.
group
Communication Manager keeps track of the last extension in the hunt group that received a
call. When another incoming call arrives, it is sent to the next idle extension, bypassing the
extension that had received the previous call.
The first extension in the hunt group will no longer be the busiest telephone while the others in
the group are sitting idle.
©2006 Avaya Inc.
Page 14
Avaya Communication Manager Feature Overview
Call Center Features
Call Center
The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of
features, capabilities, and applications designed to meet all of your customers’ call center needs.
CMS measurement of
The Call Management System (CMS) measurement of ATM feature provides the capability to
ATM
externally measure ATM trunks on CMS. The CMS messages and reports are modified to
support the expanded equipment location.
Dialed Number
This feature displays, for a called party or answering position, the service or product
Identification Service
associated with an incoming call. You administer what the system displays.
Direct agent calling
Direct agent calling lets the customer’s callers automatically go directly to the same agent
whenever they call for prompt, personalized service. These direct-to-the-agent calls are also
included in their call center measurement statistics.
Dual links to CMS
The dual links to CMS feature provides an additional TCP/IP link to a separate CMS for full,
duplicated CMS data collection functionality and high availability CMS configuration. The
same data is sent to both servers, and the administration can be done from either server.
The ACD data is delivered over different network routes to prevent any data loss from such
conditions as:
Duplicate agent login
ID administration
Agent-loginID skill
pair increase
Expert Agent
Selection
Add/remove skills
•
ACD link failures
•
CMS hardware or software failures
•
CMS maintenance
• CMS upgrades
Duplicate agent login ID administration simplifies administration of similar agent login ID
forms.
Since the LINUX platform supports 20,000 administered agent-loginIDs, the administered
agent-loginID skill pairs has been increased from 65,000 to 180,000.
With this enhancement, customers could administer an average of 9 skills per agent for the
20,000 agent-loginIDs (180,000/20,000). Customers could also administer 9,000 agents with
20 skills each (180,000/20). The number of skill pairs is administered on the Display
Capacity SAT screen using the Administered Logical Agent-Skill Pairs field. Note: This
capacity increase applies only to the S8700 Media Server and other configurations that have
the S8700 capacities. A maximum of 5,200 agents can besimultatenously active.
Expert Agent Selection (EAS) enables certain skill types to be assigned to a call type or a
Vector Directory Number (VDN). Routing calls through vectoring then allows the system
administration to direct calls to agents who have the particular agent skills required to
complete the customer inquiries.
Allows an agent using expert agent selection (EAS) to add or remove skills. A skill is a
numeric identifier that refers to the specific ability of an agent. For example, an agent who
speaks English and Spanish could be assigned a language-speaking skill with an identifier of
20. The agent then adds skill 20 to his or her set of working skills. If a customer needs a
Spanish-speaking agent, the system routes the call to an agent with that skill. Each agent can
have up to four active skills, and each skill is assigned a priority level.
©2006 Avaya Inc.
Page 15
Avaya Communication Manager Feature Overview
Call Center Features
Call Center
The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of
features, capabilities, and applications designed to meet all of your customers’ call center needs.
Call distribution based Calls that require certain agent skills (such as "knowledgeable about product X" or "speaks
on skill
Spanish") can be matched to an agent who matches the required skill. You can assign one of
up to 2,000 skill numbers to each need or group of needs. The skills are administered and
associated for each of the following:
Queue to best ISDN
support
Least Occupied Agent
Multiple call handling
(forced)
Multiple music/audio
sources
Locally sourced
announcements and
music
Multiple split queuing
Network Call
Redirection
•
Vector directory numbers (VDN)
•
Agent login IDs
• Callers
This refined skill definition capability allows you to organize call handling based on customer,
product, and language, for example.
Queue to best information is passed transparently over several public networks and QSIG
private networks using the envelopes that are part of the QSIG Manufacturer-Specific
Information (MSI) and the ISDN platform enhancement.
The Least Occupied Agent (LOA) feature distributes calls evenly across all available agents,
balancing the workload among agents with fewer skills and agents with several skills. LOA
solves the problem of agents who are bombarded with calls after logging into a skill at the
start of a shift, while the agents who are already logged in have maintained their current
incoming call level.
This feature allows agents to receive an ACD call while other types of calls are alerting,
active, or on hold.
Multiple music/audio sources lets customers deliver music or customized announcements to
callers while they are in queue, helping to make the waiting time more productive or
entertaining. Customers can provide information about their products, services, other call
center applications, offer public service information, or play music.
Use the Locally Sourced Announcements and Music feature to access announcement and
music audio sources on a local port network or media gateway.
Locally sourced audio can:
•
improve the quality of audio
•
reduce resource usage, such as VoIP resources
• provide a backup mechanism for announcement and music sources
Multiple split queuing lets customers direct a call to several splits at the same time, so that the
first available agent can take the call. It can help customers handle the busiest periods with
greater ease and provide faster service to their callers.
Today, call center customers are looking for many ways to reduce their costs. One of these
ways is to employ Public Switched Telephone Network (PSTN) virtual private networks
(VPNs) to eliminate as much private network cost as possible. These cost reductions are
particularly valuable in enterprises or multi-site call-center environments and especially to
enterprise call centers where network costs are typically high.
Network call redirection (NCR) offers a call redirection method between sites on a public
network or a PSTN VPN, to help reduce trunking costs. NCR may only be activated for
incoming ISDN trunk calls where the associated trunk group has been enabled by the public
network service provider to use network call transfer or network call deflection features.
©2006 Avaya Inc.
Page 16
Avaya Communication Manager Feature Overview
Call Center Features
Call Center
The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of
features, capabilities, and applications designed to meet all of your customers’ call center needs.
ETSI Explicit Call
The Network Call Redirection (NCR) support of the "ETSI Explicit Call Transfer" feature is
Transfer signaling
desired by multi-site, non-U.S. Avaya call center customers who use various PSTN service
providers for ISDN services. These non-U.S. call centers wish to accomplish call transfers
between sites without holding the ISDN trunks of a transferred call at the call redirecting
Communication Manager site.
The Network Call Redirection/Network Call Deflection (NCR/NRD) feature does not allow for
announcement and call-prompting call-vectoring operations. Therefore, the ETSI ECT feature
is for these call center customers who cannot use NCR/NRD since they wish to play an
announcement to a caller and use Communication Manager call-prompting to allow the caller
to determine the routing for the call.
Network call
This enhancement adds support for the 2B-Channel Transfer PSTN network transfer
redirection 2B-channel protocols to the Network Call Redirection (NCR) feature. The protocols that are supported
transfer
are:
•
PC Application
Software Translation
Exchange
Priority queuing
Reason codes
Redirection on no
answer
Remote logout of
agent
Service observing
Service observing by
COR
Telcordia TBCT (offered by local and inter-exchange PSTNs with Lucent 5Ess or
Nortel DMS100 switches in US or Canada)
• 1998 ANSI Explicit Call Transfer (ECT) for future use
Another form of network transfer is where the PBX sets up the second leg call and asks the
network to merge the incoming call with the outgoing call (the 2B-channels) and drops the
trunks to the PBX.
PC Application Software Translation Exchange (PASTE) allows users to view call center data
on display telephones, displaying what each terminal button is, and what the feature access
codes for the switch are. PASTE is used in conjunction with Avaya IP agent.
Priority queuing allows special callers to be assigned priority status and routed ahead of other
callers. Clients can pamper their best customers with the fastest attention possible.
Allows agents to enter a numeric code that describes their reason for entering auxiliary (AUX)
work mode or for logging out of the system. Reason codes give call center managers detailed
information about how agents spend their time. You can use this data to develop more
precise staffing forecasting models or use it with schedule-adherence packages to ensure that
agents are performing scheduled activities at the scheduled time. You must have expert agent
selection (EAS) enabled to use reason codes.
This feature redirects a ringing ACD split or skill call or direct agent call after an administered
number of rings. This prevents an unanswered call from ringing indefinitely. The call can
redirect either to the split or skill to be answered by another agent or to a Vector Directory
Number (VDN) for alternative call handling. Direct agent calls route to the agent coverage
path, or to a VDN if no coverage path is administered. You must have ACD enabled to use
this feature.
The remote logout of agent feature allows a select set of users to log out an agent using a
feature access code.
Service observing allows a specified user, such as a supervisor, to observe or monitor calls of
another user. A vector directory number call can also be observed. Observers can observe in
listen-only or listen-and-talk mode. You set up service observing to observe a particular
extension, not all calls to all extensions at a terminal. Note: Service observing may be subject
to federal, state, or local laws, rules, or regulations or require the consent of one or both of the
call parties. Familiarize yourself and comply with all applicable laws, rules, and regulations
before using this feature.
Service observing by class of restriction (COR) restricts certain users from using the service
observing feature.
©2006 Avaya Inc.
Page 17
Avaya Communication Manager Feature Overview
Call Center Features
Call Center
The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of
features, capabilities, and applications designed to meet all of your customers’ call center needs.
Service observing of
Service observing of VDNs (also known as VDN observing on agent answer) allows a
VDNs
supervisor to start observing a call to the VDN when the call is delivered to the agent station.
The observer will not hear the call during vector processing (announcements, music, and so
on).
Service observing
This option will allow observing from non-feature button equipped stations. An observer will be
remote
able to monitor a VDN or a physical extension remotely using an "observe FAC" procedure
through the remote access feature and/or call vectoring/call prompting features (through
VDNs).
Vector-initiated
Vector-initiated service observing, also called VDN observing on agent answer, allows users
service observing
to start observing of a call to the VDN when the call is delivered to the agent or station. This
saves time for the observer after observing of the VDN has been activated since the observer
does not have to wait listening for each subsequent call to go through vector processing and
for the agent to answer.
Listen-only FAC for
The system provides a no-talk, listen-only service observing feature access code (FAC). This
service observing
FAC does not reserve a second timeslot for potential toggle to talk and listen mode. This
feature is for call recording applications that use Service Observing of stations/ACD agents to
provide increased call recording capacity by reducing the timeslot usage
Site statistics for
The site statistics for remote port networks feature forwards location IDs to CMS to provide
remote port networks
call center site-specific reports.
User-to-user
This feature provides the mechanism to pass information across several key public networks,
information over the
including information that is originated or destined for one of several applications on
public network
Communication Manager.
Voice Response
Voice Response Integration (VRI) integrates call vectoring with the capabilities of voice
Integration
response units such as the Avaya CONVERSANT voice information system. You can also
integrate a voice response unit with ACD. All this provides a variety of advantages. For
example, while a call is queued, a caller can listen to product information via an audiotext
application or can complete an interactive voice-response transaction. It may be possible to
resolve the caller questions while the call is queued, which helps reduce queuing time for
other callers during peak times.
VuStats
VuStats presents BCMS statistics on telephone displays. Agents, supervisors, call center
managers, and other users can press a button and view statistics for agents, splits or skills,
VDNs, and trunk groups. These statistics can help agents monitor their own performance, or
respond appropriately to the caller request. Features include:
•
VuStats login IDs
•
VuStats service level
©2006 Avaya Inc.
Page 18
Avaya Communication Manager Feature Overview
Collaboration Features
Collaboration
Avaya Communication Manager contains a variety of features aimed at providing easy ways to collaborate with groups of
peers, customers, and partners such as executives, sales people, and professional specialists. These key work groups
require a high level of effective interaction, and Communication Manager delivers.
Conferencing
Abort conference on
When you press the conference button and for any reason you hang up before you complete
hang-up
the conference, you will cancel the conference. The original call that was put on soft-hold is
put on hard-hold.
Conference -three
The conference button allows single-line telephone users to make up to three-party
party
conference calls without attendant assistance.
Conference -six party
The conference button allows multi-appearance telephone users to make up to six-party
conference calls without attendant assistance.
Conference/transfer
Conference/transfer display prompts are based on the user class of restriction (COR). The
display prompts
display prompts are based on the user COR, independent of the select line appearance
conferencing and no dial tone conferencing feature. The display messages vary depending on
the activation of the two features, but the choice of displaying the additional information or not
is dependent on the station user COR.
Conference/transfer
The conference/transfer toggle/swap feature allows users to toggle between two parties in the
toggle/swap
middle of setting up a conference call prior to connecting all parties together, or to consult with
both parties prior to transferring a call. The display also toggles between the two parties.
Group listen
The group listen feature simultaneously activates your speakerphone in listen-only mode, and
your handset or headset in listen-and-speak mode. This allows you to serve as spokesperson
for a group. You can participate in a conversation while everyone else in the room is listening
to what is said. Note: This feature works only on certain types of telephones. It is not
supported on IP telephones.
Hold/unhold
Allows user to use the Hold button to bring the held party back to the conversation. This is an
conference
alternative to using the line appearance button. Hold/unhold only applies if there is only one
line on hold and no other line appearances are active. An error message is displayed if the
unhold feature is attempted when not allowed. Note: This feature is not available for BRI
stations or attendant consoles.
Meet-me Conferencing The Meet-me Conferencing feature allows a person to set up a dial-in conference of up to six
parties. The Meet-me Conferencing feature uses call vectoring to process the setup of the
conference call.
Meet-me Conferencing can be optionally set up to require an access code. If an access code
is assigned, and if the vector is programmed to expect an access code, each user dialing in to
the conference call must enter the correct access code to be added to the call.
The Meet-me Conferencing extension can be dialed by any internal or remote access users,
and by external parties if the extension number is part of the customer DID block.
Expanded Meet-me
Use the Expanded Meet-me Conferencing application to set up multi-party conferences
Conferencing
consisting of more than six parties. The Expanded Meet-me Conferencing application
supports up to 300 parties. This application is available with Communication Manager release
3.0 or later.
The Expanded Meet-me Conferencing application requires an external Meeting Exchange
(MX) server.
No dial tone
This feature can eliminate user confusion over receiving dial tone when trying to conference
conferencing
two existing calls. It skips the automatic line selection if there is already a party on hold or an
alerting line appearance. Help messages help guide the user. This feature is assigned on a
system wide basis.
©2006 Avaya Inc.
Page 19
Avaya Communication Manager Feature Overview
Collaboration Features
Collaboration
Avaya Communication Manager contains a variety of features aimed at providing easy ways to collaborate with groups of
peers, customers, and partners such as executives, sales people, and professional specialists. These key work groups
require a high level of effective interaction, and Communication Manager delivers.
No hold conference
This feature allows a user to automatically add another party to a conference call while
continuing the conversation of the existing call. The new party is automatically entered into
the conversation as soon as the call is answered. An optional tone can be provided prior to
the party being added to the call. Note: The calling station cannot hold, conference, or transfer
an Emergency Access to Attendant call. This applies to both the traditional means of using
these features, and to the no-hold method of using these features.
After dialing is complete, if the No Hold Conference is not answered within the time specified
in an administered "timeout" field, the No Hold Conference call is deactivated.
Select line appearance If you are in a conversation on line "b", and another line is on hold or an incoming call is
conferencing
alerting on line "a", then pressing the CONF button bridges the calls together. Using the select
line appearance feature on Communication Manager, the user has the option of pressing a
line appearance button to complete a conference instead of pressing CONF a second time.
This feature only applies if the line is placed in soft hold by pressing the CONF button. This
feature never applies if the soft hold was due to pressing a TRANSFER button.
Selective conference
The selective conference party display, drop, and mute feature allows any user on a digital
party display, drop,
station with display or on an attendant console to use the display to identify all of the other
parties on a two-party or conference call.
and mute
The user would press a feature button while on the call that puts the station or console into
conference display mode. The user then can scroll through the display of each party currently
on the call by repeatedly pressing the feature button. The display would show the number and
name (when available) of the caller.
The user could then do either of the following:
•
The user can selectively drop the party currently shown on the display with a single
button push. This can be useful during conference calls when adding a party that
does not answer and the call goes to voice mail.
•
Selective conference
mute
Multimedia calling
The user can selectively mute the party currently shown on the display with a single
button push. This puts the selected party in "listen-only" mode. This can be useful
during conference calls when a party puts the conference call on hold and everyone
on the call is forced to listen to music-on-hold. The user can mute that party so the
conference call can continue without interruption. The muted party can then rejoin
the call by pressing the # key on their telephone.
Selective conference mute allows a conference call participant, who has a display station, to
mute a noisy trunk line. Selective conference mute is also known as far end mute.
Examples of noisy trunk lines that might need to be muted during a conference call are:
•
cell telephones
•
telephones that utilize the Music-On-Hold feature
• telephones with no mute capabilities
Selective conference mute only applies to trunk lines on the conference call, and not to
stations. Only one trunk line on the conference call can be selectively muted at a time. This
enhanced conferencing feature can be activated from any display station with a "conf-dsp"
button and an "fe-mute" button.
Multimedia calls are initiated with voice and video only. Once a call is established, one of the
parties may initiate an associated data conference to include all of the parties on the call who
are capable of supporting data. The data conference is controlled by an adjunct device called
an Expansion Services Module (ESM).
©2006 Avaya Inc.
Page 20
Avaya Communication Manager Feature Overview
Collaboration Features
Collaboration
Avaya Communication Manager contains a variety of features aimed at providing easy ways to collaborate with groups of
peers, customers, and partners such as executives, sales people, and professional specialists. These key work groups
require a high level of effective interaction, and Communication Manager delivers.
Multimedia
The multimedia Application Server Interface (ASA) provides a link between Communication
Application Server
Manager and one or more multimedia communications eXchange nodes. A Multimedia
Interface
Communications Exchange (MMCX) is a stand-alone multimedia call processor produced by
Avaya. This link to Communication Manager enhances the capabilities of each multimedia
communications eXchange system by enabling it to share some of the Communication
Manager features.
In particular, the interface provides the following advantages:
•
Call Detail Recording (CDR) -This allows you to capture call detail records so you
can analyze the call patterns and usage of multimedia calls just as Communication
Manager Administrators analyze normal calls.
•
Automatic Alternate Routing/Automatic Route Selection (AAR/ARS) -This allows for
the intelligent selection of the most cost-effective routing for calls, based on available
resources and your carrier preference. The system may select public trunks through
a DEFINITY® MultiMedia Communication Exchange (MMCX).
•
Multimedia call early
answer on vectors and
stations
Voice mail integration -You can access your embedded AUDIX or INTUITY AUDIX
voice messaging system from a MultiMedia Communication Exchange (MMCX).
Early answer is a feature applied to multimedia calls in conjunction with conversion to voice.
The early answer feature:
Answers the data call
•
Establishes the multimedia protocol prior to completion of a converted call
•
Multimedia Call
Handling
Multimedia call
redirection to
multimedia endpoint
Multimedia data
conferencing (T.120)
through an ESM
Multimedia hold,
conference, transfer,
and drop
Multimedia queuing
with voice
announcement
Ensures that a voice path to/from the originator is available when the voice call is
answered
For an incoming call, early answer answers the dynamic service-link calls when the
destination endpoint answers, unless early answer is specified during routing or termination
processing. Note: The "destination voice endpoint" might be an outgoing voice trunk if the
destination voice station is forwarded or covered off-premises.
See Multimedia Call Handling.
A dual port multimedia station may be a destination of call redirection features such as call
coverage, forwarding, and station hunting. The station can receive and accept full multimedia
calls or data calls converted to multimedia.
The data conference is controlled by an adjunct device called an Expansion Services Module
(ESM). The ESM is used to terminate T.120 protocols [including Generalized Conference Call
(GCC), a protocol standard for data conference control] and provide data conference control
and data distribution. The MultiMedia Interface circuit pack, TN787, is used to rate adapt
T.120 data to/from the ESM.
Station users have the ability to activate hold, conference, transfer, or drop on multimedia
calls. Multimedia endpoints and voice-only stations may participate in the same conference.
When multimedia callers queue for an available member of a hunt group, they are able to
hear an audio announcement.
©2006 Avaya Inc.
Page 21
Avaya Communication Manager Feature Overview
Collaboration Features
Collaboration
Avaya Communication Manager contains a variety of features aimed at providing easy ways to collaborate with groups of
peers, customers, and partners such as executives, sales people, and professional specialists. These key work groups
require a high level of effective interaction, and Communication Manager delivers.
Avaya Video
The Avaya Video Telephony Solution makes video calls as simple and easy as a regular
Telephony Solution
telephone call. The Avaya Video Telephony Solution is fully integrated into your standard dial
plan, enabling totally transparent and seamless voice and video conferencing, both for the
desktop and for group video communications.
Communication Manager features such as hold, transfer, resume, and conference are
seamless with video conferencing adjuncts from Polycom. Avaya Video Telephony Solution
unifies Voice over IP with video, web applications, Avaya’s video enabled IP Softphone, third
party gatekeepers, and other H.323 endpoints.
The following components are part of the Avaya Video Telephony Solution feature:
•
Polycom VSX3000, VSX7000, and VSX 8000 conferencing systems with Release
8.03 or later
•
Polycom V500 video calling systems
•
Polycom MGC video conferencing bridge platforms with Release 7.02
• Third party gatekeepers
The solution requires Communication Manager Release 3.0.1, and Avaya IP Softphone
release 5.2, with Avaya Integrator for Polycom Video release 2.0.1.
The Avaya Video Telephony Solution also supports the:
•
Logitech 4000 Pro web camera
•
Polycom Via Video
•
Code calling access
Group paging
Intercom -automatic
Intercom -automatic
answer
Intercom -dial
Creative Labs notebook webcam
Paging and intercom
This feature allows attendants, users, and tie trunk users to page with coded chime signals.
This feature is helpful for users who are often away from their telephones or at a location
where a ringing telephone might be disturbing.
Group paging allows a user to make an announcement to a group of people using
speakerphones. The speakerphones are automatically turned on when the user begins the
announcement. The recipients can listen to the message over the handset if they wish, but
they cannot speak to the user in return. A group page member will not receive the page if the
member is active on a call appearance, has a call ringing, is off-hook, has "send-all calls"
active, or has "do not disturb" active.
With this feature, users who frequently call each other can do so by pressing one button
instead of dialing an extension number. Calling users press the automatic intercom button and
lift the handset. The called user receives a unique intercom ring and the intercom lamp, if
provided, flashes.
Automatic answer intercom (auto answer ICOM) allows a user to answer an intercom call
within the intercom group without pressing the intercom button. Auto answer ICOM works with
digital, BRI, and hybrid telephones with built-in speaker, headphones, or adjunct
speakerphone.
This feature allows multi-appearance telephone users to easily call others within an
administered group. The calling user lifts the handset, presses the dial intercom button, and
dials the one-digit or two-digit code assigned to the desired party. The telephone of the called
user rings, and the intercom lamp, if provided, flashes. With this feature, a group of users who
frequently call each other can do so by pressing one button and dialing a one-digit or two-digit
code instead of dialing an extension number.
©2006 Avaya Inc.
Page 22
Avaya Communication Manager Feature Overview
Collaboration Features
Collaboration
Avaya Communication Manager contains a variety of features aimed at providing easy ways to collaborate with groups of
peers, customers, and partners such as executives, sales people, and professional specialists. These key work groups
require a high level of effective interaction, and Communication Manager delivers.
Loudspeaker paging
Loudspeaker paging access provides attendants and telephone users dial access to voice
access
paging equipment. As many as nine paging zones can be provided by the system, and one
zone can be provided that activates all zones at the same time. Note: A zone is the location of
the loudspeakers -for example, conference rooms, warehouses, or storerooms.
A user can activate this feature by dialing the trunk access code of the desired paging zone,
or the access codes can be entered into abbreviated dialing lists. Once you have activated
this feature, you can simply speak into the handset to make the announcement.
Deluxe loudspeaker paging access (called deluxe paging) provides attendants and telephone
users with integrated access to voice-paging equipment and call park capabilities. When you
activate deluxe paging, the call is automatically parked. The parked call returns to the parking
user with distinctive alerting when the time-out interval expires.
Manual signaling
Allows one user to signal another user. The receiving user hears a two-second ring. The
signal is sent each time the button is pressed by the signaling user. The meaning of the signal
is prearranged between the sender and the receiver. Manual signaling is denied if the
receiving telephone is already ringing from an incoming call.
Whisper page
Whisper page allows an assistant or colleague to bridge onto your telephone conversation
and give you a message without being heard by the other party or parties you are talking to.
Whisper page works only on certain types of telephones.
©2006 Avaya Inc.
Page 23
Avaya Communication Manager Feature Overview
Communication Device Support Features
Communication Device Support
Communication Manager supports Avaya’s complete portfolio of analog, digital, IP, SIP, hardphone, softphone, wireless,
and personal user agent solutions.
Avaya IP Agent
Avaya IP Agent is a PC-based IP application that allows agents to use their PCs as
telephones. In addition to the traditional functionality of a standard telephone (transfer, hold,
conference, and so forth), IP agent offers directory services, screen pops, call history, and
agent mode history.
Avaya IP Softphone
Avaya IP Softphone extends the level of Communication Manager services. This feature turns
a PC or a laptop into an advanced telephone. Users can place calls, take calls, and handle
multiple calls on their PCs. Note: R1 and R2 IP Softphone and IP Agent, which use a dual
connect (two extensions) architecture, are no longer supported. R3 and R4 IP Softphone and
IP Agent, which use a single connect (one extension) architecture, continue to be supported.
This applies to the RoadWarrior configuration and the Native H.323 configuration for the IP
Softphone.
The R5 release of the IP Softphone supports a number of enhanced features, including the
following:
IP Softphone and IP
Agent -RoadWarrior
mode
IP Softphone and IP
Agent -Shared Control
mode
IP Softphone and IP
Agent -Telecommuter
mode
•
Improved endpoint connection recovery algorithm
•
AES media encryption
•
Instant Messaging
•
Unicode support
• Softphone and Telephone Shared Control
The IP Softphone provides a graphical user interface with enhanced capabilities when used
with certain models of DCP telephones. Communication Manager supports a mode of H.323
registration that allows an IP Softphone to register for the same extension as a DCP
telephone without disabling the telephone. It also allows the IP Softphone to send button-push
messages and receive display and call progress messages in parallel with the telephone. In
this mode, the Softphone does not terminate any audio.
IP Softphone and IP Agent, RoadWarrior mode, enables use of the full Avaya Communication
Manager feature set from temporary remote locations anywhere in the world. The
RoadWarrior application consists of two software applications running on a PC that is
connected to Communication Manager over an IP network.
The single network connection between the PC and Communication Manager carries two
channels, one for the signaling path and one for the voice path. On Communication Manager,
the RoadWarrior application requires the CLAN circuit pack for signaling and the IP media
processor for voice processing.
IP Softphone and IP Agent, Shared Control mode, enables users to have a telephone
endpoint and an IP Softphone in service simultaneously on the same extension number. IP
Softphone and an IP telephone can be integrated so that the IP softphone can control a desk
IP telephone. This allows the power of the PC desktop (LDAP directories, TAPI PIMs/Contact
Managers, etc.) to be used in conjunction with a desktop IP telephone.
An IP softphone can register to an extension number that is already assigned to an in-service
telephone endpoint. From that point on, user actions carried out by either endpoint apply to
calls to or from the extension. Only the telephone endpoint carries audio for the extension,
however.
IP Softphone and IP Agent, Telecommuter mode, enables telecommuters to use the full
Communication Manager feature set from home. It consists of a PC and a telephone with
separate connections to Communication Manager. The PC provides the signaling path and
the user interface for call control. A standard telephone provides a high-quality voice path.
The Telecommuter application requires the CLAN circuit pack for signaling. The
Telecommuter application does not use the IP media processor.
©2006 Avaya Inc.
Page 24
Avaya Communication Manager Feature Overview
Communication Device Support Features
Communication Device Support
Communication Manager supports Avaya’s complete portfolio of analog, digital, IP, SIP, hardphone, softphone, wireless,
and personal user agent solutions.
Avaya IP Softphone
Avaya IP Softphone for pocket PC extends the level of Communication Manager services.
for pocket PC
This feature turns a hand-held personal digital assistant (PDA) into an advanced telephone.
Users can place calls, take calls, and handle multiple calls on their PDAs.
Avaya Communication The Communication Manager PC console allows your attendants to efficiently handle
Manager PC console
incoming calls by personal computer. Using the familiar Microsoft Windows graphical user
interface (GUI), the attendants can easily keep track of how long callers have been on hold
and who they are waiting for. Attendants can monitor up to six calls at once.
Attendants do not need to use pen and paper when handling calls because they can make
notes on their computers about what each caller needs. All this contributes to make a
favorable first impression with your customers. Having the call processing software on the
same computer with spreadsheet, word processing, or other software allows the attendants to
stay productive between calls.
The PC console is easily customized, so even if attendants from different shifts share the
same computer, they can each preserve their preferences in the call processing environment.
The PC console is available in English, Parisian French, Latin American Spanish, German,
Dutch, Italian, and Portuguese. If a Spanish-speaking attendant takes over for a Frenchspeaking attendant, for example, a single press of a button converts all labels, error
messages, and online help to Spanish.
Avaya SoftConsole
The Avaya SoftConsole is a Windows-based GUI application that can replace the physical
302B "hard" console. It allows attendants to perform call answering and routing through a PC
interface through an IP connection.
Unicode support
Communication Manager supports the display of non-English static and dynamic display text
on Unicode-enabled telephones. Non-English display information is entered into a Avaya
Integrated Management application. Communication Manager processes, stores, and
transmits the non-English text to telephones that support Unicode displays.
Unicode support provides the capability of supporting international and multi-national
communications solutions. End-users are provided with a communications interface (delivered
by an IP telephone or IP Softphone) in their own native language. This feature supports the
Simplified Chinese, Japanese, and Korean (CJK) character sets.
QSIG support for
The QSIG support for Unicode feature extends the Unicode support on a single server to
Unicode
multi-node Communication Manager networks. This feature allows Unicode support across
large campus configurations. Many configurations contain multiple Communication Manager
servers due to scalability requirements. This feature also allows Unicode support across large
corporate networks, frequently multinational corporations, where multiple Communication
Manager servers are almost always provisioned.
More simultaneous
Communication Manager can handle multipoint endpoints that are capable of up to six calls at
calls per multipoint
once.
endpoint
Direct-region
In many customer configurations, IP telephones are placed into their own direct network
regions or indirect network regions. Voice over IP (VOIP) allocation can now favor directpreference for IP
telephones
connected regions over indirect connected regions.
©2006 Avaya Inc.
Page 25
Avaya Communication Manager Feature Overview
Hospitality Features
Hospitality
Alphanumeric dialing
Attendant room status
Automatic selection of
Direct Inward Dialing
numbers
Automatic wakeup
Check-in/check-out
Custom selection of
VIP DID numbers
Daily wakeup
Dial-by-name
Do not disturb
Alphanumeric dialing allows you to place data calls by entering an alphanumeric name rather
than a long string of numbers.
See Attendant room status.
This feature allows the system to automatically choose a number from a list of available Direct
Inward Dialing (DID) numbers that will be assigned to a guest room extension when checking
in.
With this feature, hotels can give a guest a second telephone number that is different from
their room number, thereby protecting the privacy of the guest. When a particular DID number
is called, the call routes to the guest room extension, and covers as if the room was called
directly. Besides improving guest security, this eliminates the need for an attendant or front
desk staff to extend a call to a guest room.
The automatic wakeup feature allows attendants, front desk users, and guests to request that
one or two wake-up calls be automatically placed to a certain extension number at a later
time. When a wakeup call is placed and answered, the system can provide a recorded
announcement (which can be a speech synthesis announcement), music, or simply silence.
With the integrated announcement feature, multiple announcements enable international
guests to use wakeup announcements in a variety of languages.
This feature allows front desk personnel to check guests into a hotel and, when the guests
leave, check them out. There are two ways this is done: through the PMS terminal or through
the attendant console (or backup telephone). Check-in and check-out from the attendant
console should be used only if there is no Property Management System (PMS), or if the link
to the PMS is down. If the PMS is installed and working, check guests in and out using the
PMS.
For guest check-in or check-out from the console, there are two buttons on the attendant
console (or backup telephone): one labeled "Check in" and the other labeled "Check out." The
check-in procedure performs two functions: it deactivates the restriction on the telephone in
the room allowing outward calls, and it changes the status of the room to occupied.
This feature builds on the automatic selection of DID numbers feature. It allows hotel
personnel to control what DID number is assigned to a hotel room at check-in. That is, the
system asks the user to specify the desired DID number when a guest is checked in. The
number comes from a pool of DID numbers that are separate from those used by the
automatic selection feature. The system never automatically assigns numbers from this pool.
Numbers from this pool are used only when explicitly specified by the user.
Daily wakeup allows a guest or front desk personnel to schedule a single wakeup request for
a daily wakeup call. For example, if a guest needs to receive a wakeup call at 5:30 a.m. for
the duration of his or her stay, one request can be placed on the system instead of placing a
separate request for each day.
The dial-by-name feature allows callers to the system to access guest rooms simply by dialing
the name of the guest they are trying to contact. This feature uses recorded announcements
and the call vectoring feature to set up an automatic attendant procedure. This automatic
attendant procedure gives callers the ability to enter a guest name. When a single or unique
match is found, the call is redirected to the telephone of the guest.
The do not disturb feature allows guests, attendants, and authorized front desk users to
request that no calls, other than priority calls, be connected to a particular extension until a
specified time.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Hospitality Features
Hospitality
Dual wakeup
Housekeeping status
Names registration
Property Management
System digit to
insert/delete
Property Management
System interface
Single-digit dialing
and mixed station
numbering
Suite check-in
VIP wakeup
Wake-up activation
using confirmation
tones
This feature allows guests to have two separate wakeup calls. The dual wakeup feature is an
enhancement to the standard automatic wakeup feature used in hospitality environments.
With the standard wakeup feature, guests or front desk personnel can create one wakeup call
for each extension. The dual wakeup feature allows guests and front desk personnel to create
either one or two wakeup calls. The dual wakeup feature for guests is valid only when the
system is not equipped with a speech synthesizer circuit pack.
The housekeeping status feature records the status for up to six housekeeping codes and
reports them to the property management system (PMS). These status codes are usually
entered by the housekeeping staff from the guest room or from a designated telephone. They
can also be updated by the front office personnel using the attendant console or a backup
telephone. Six status codes can be used from guest rooms, and four status codes can be
used from telephones that do not have the client room class of service (COS).
The names registration feature automatically sends a guest name and room extension from
the property management system (PMS) to the switch at check-in, and automatically removes
this information at check-out. The information may be displayed on any attendant console or
display-equipped telephone at various hotel locations (for example, room service or security).
Many customer configurations base a room extension by adding an extra leading digit on the
room number. The PMS digit to insert/delete feature allows users to delete the leading digit of
the extension in messages. The feature is useful for a hotel that has multiple extensions
sharing an extra leading digit in front of the room number. The leading digit is automatically
inserted when the message goes to the switch.
The PMS interface supports 3-digit, 4-digit, or 5-digit extensions, but prefixed extensions do
not send the entire number across the interface. Only the assigned extension number is sent.
Therefore, you should not use prefixed extensions for numbers that are also going to use the
digit to insert/delete function.
The Property Management System (PMS) allows a customer to control features used in both
a hospital-type and a hotel/motel-type environment. The communications link allows the
property management system to interrogate the switch, and allows information to be passed
between the switch and the PMS. The switch exchanges guest status information (room
number, call coverage path, and other data) with the PMS.
This feature provides hotel staff and guests easy access to internal hotel/motel services, and
provides the capability to associate room numbers with guest room telephones. The feature
provides the following dial plan types: single-digit dialing, prefixed extensions, and mixed
numbering.
Suite check-in allows more than one station to be checked in at one time. This is useful for a
guest room that may have multiple extensions. This feature allows all extensions to be
checked in at the same time. Suite check-in using the hunt-to feature will also check out all
the extensions in the entire suite at the same time.
The VIP wakeup feature allows front desk personnel to provide personalized wakeup calls to
important guests. When a wakeup call has been scheduled for an important guest, a wakeup
reminder call is placed to the front desk personnel, who in turn personally calls the guest to
provide the wakeup call.
If a speech synthesizer circuit pack is not installed, guests can still enter their own wakeup
calls (two wakeup calls if the dual wakeup feature is active). The guests do not receive voice
prompts as they would using the speech synthesizer circuit pack. Instead, guests receive call
progress tones (recall dial tone and confirmation tone) to set up their wakeup calls.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Hospitality Features
Hospitality
Xiox call accounting
The Xiox call accounting works as an adjunct with any system with hospitality features. Xiox
call accounting allows hotel management to use their telephone system as a major source of
revenue by generating the information they need to make important decisions about their
network and usage.
©2006 Avaya Inc.
Page 28
Avaya Communication Manager Feature Overview
Localization Features
Localization
Communication Manager offers a variety of features supporting your global enterprise.
Administrable
This feature allows messages that appear on telephone display units to be shown in the
language displays
language spoken by the user. These messages are available in English (the default), French,
Italian, Spanish, or one other user-defined language. The language for display messages is
selected by each user. The feature requires 40-character display telephones.
Administrable loss
The administrable loss plan provides the ability to administer signal loss and gain for
plan
telephone calls. This capability is necessary because the amount of loss allowed on voice
calls can vary by country. With the administrable loss plan feature, switch endpoints are
classified into 17 endpoint types, and the loss plan can be administered for trunks, stations,
and personal CO lines. Loss values are in the range of 15 dB loss to 3 dB gain. Preset
defaults are available and are based on country type.
Bellcore calling name
This feature allows the system to accept calling name information from a Local Exchange
ID
Carrier (LEC) network that supports the Bellcore calling name specification. The system can
send calling name information in the format if Bellcore calling name ID is administered. The
following caller ID protocols are supported:
•
Bellcore (default) - US protocol (Bellcore transmission protocol with 212 modem
protocol)
•
Block collect call
Busy tone disconnect
Distributed
Communications
Systems protocol
V23-Bell - Bahrain protocol (Bellcore transmission protocol with V.23 modem
protocol).
This feature blocks collect calls on class-of-restriction basis. This feature is available for any
switch that uses the Brazil country code. If enabled for a station, all trunk calls that terminate
to the station will send back a double answer to the central office (CO). This double answer
tells the CO that this particular station cannot accept collect calls. The CO then tears down
the call if it is a collect call.
In some regions of the world, the CO sends a busy tone for the disconnect message. With
busy tone disconnect, the switch disconnects analog loop-start CO trunks when a busy tone is
sent from the CO.
Country-specific localization
Italy
Enhanced DCS adds features to the existing DCS capabilities and requires the use of Italian
TGU/TGE tie trunks.
Additional features include:
•
Exchanging information to provide class of restriction (COR) checking between
switches in the EDCS network
•
Providing call-progress information for the attendant
•
Allowing attendant intrusion between a main and a satellite PBX
•
National private
networking support
Katakana character
set
Allowing a main PBX to provide DID/CO intercept treatment rather than the satellite
PBX
Japan
Provides support for Japanese private ISDN networks. The Japanese private network ISDN
protocol is different from the standard ISDN protocol. The switch supports extensions to the
ISDN protocol for switches using the Japanese country code.
Communication Manager supports the katakana character set (Japan). This nine-point
character font was designed to display katakana characters in the user interface as well as in
switch-generated messages.
©2006 Avaya Inc.
Page 29
Avaya Communication Manager Feature Overview
Localization Features
Localization
Communication Manager offers a variety of features supporting your global enterprise.
Called number added
Depending on how you set the Outgoing Display field on the Trunk screen, a call from a
to display for Toshiba Toshiba SIP telephone over a non-ISDN trunk, to which another Toshiba SIP telephone is
SIP telephone
added, now displays either the trunk name or the dialed number.
Setting the Outgoing Display field to Yes displays the trunk name. Setting the Outgoing
Display field to No displays the dialed number.
Russia
Central Office support Communication Manager supports central office (CO) trunks in Russia using the G700 Media
on G700 Media
Gateway.
Gateway
ISDN/DATS network This feature supports ISDN/DATS trunk networks when the tone generated field is set to 15
support
(Russia) on the system-parameters country-options screen. It modifies the overlap sending
delay and ISDN T302 and T304 timers to support the Russian trunk network.
Multi-Frequency
Multi-Frequency Packet (MFP) address signaling is provided in Russia on outgoing CO
Packet signaling
trunks. Calling party number and dialed number information is sent on outgoing links between
local and toll switches. Russian MFP is set on each trunk group on the type field on the trunk
screen. Note: Russian MFP does not apply to PCOL trunks.
E&M signaling Continuous and pulsed E&M signaling is a modification to the E&M signaling used in the
continuous and
United States. Continuous E&M signaling is intended for use in Brazil, but can also be used in
Hungary. Pulsed E&M signaling is intended for use in Brazil.
pulsed
Multinational
For customers who operate in more than one country, the Multinational Locations feature
Locations
provides the ability to use a single Enterprise Communication Server (ECS) across multiple
countries.
The S8300, S8500, and S8700 Media Server each supports 25 location parameter sets. You
can administer one parameter set for each country that you support, for a maximum of 25
countries. Note: Since the S8100 Media Server supports only 1 location, and since the
Multinational Locations feature depends on multiple locations, the Multinational Locations
feature is not supported on the S8100 platform.
Analog line board
You can administer the following analog line board parameters for each location:
parameters per
• Analog Ringing Cadence
location
• Analog Line Transmission
Companding for
DCP telephones and
circuit packs per
location
•
Flashhook Interval Upper Bound
•
Flashhook Interval Lower Bound
•
Forward Disconnect Timer (msec)
• Analog line tests use the same parameters
Analog line circuit packs use these parameters, according to the location parameters of the
circuit pack.
You can administer the Companding Mode for each remote office, media gateway, and the
rest of the system that is circuit switched.
•
When a Digital Communications Protocol (DCP) telephone comes into service,
Communication Manager downloads the correct companding mode for the location
of the telephone.
•
When a circuit pack comes into service, Communication Manager downloads the
administered companding mode for the media server, remote office, or media
gateway that is supporting that circuit pack.
©2006 Avaya Inc.
Page 30
Avaya Communication Manager Feature Overview
Localization Features
Localization
Communication Manager offers a variety of features supporting your global enterprise.
Location ID in Call
You can administer the following CDR parameters in the custom CDR format for both the
Detail Record
source and destination:
records
• location
•
Loss plans per
location
Multifrequency
signaling per trunk
group
Tone generation per
location
Public network call
priority
World class tone
detection
time zone
• country
For each location, you can administer Digital Loss & Tone Loss, DCP terminal loss
parameters, and administrator-entered customizations.
When inserting loss for a multilocation intrasystem call, Communication Manager treats the
call as if IP tie trunks are connecting the different parties. When an audio stream is converted
from time-division multiplexing (TDM) to Internet Protocol (IP), the system adjusts the audio
stream. The system adjusts the audio stream by the IP media processor board of the sending
location, to an ISO standard level for voice over IP. The system then adjusts the audio stream
by the media processor of the receiving location to match the TDM levels for that location.
This board level adjustment is not done for DS1 remoted expansion port networks (EPNs).
Use DS1 remoted EPNs between countries only if the countries have similar voice transmit
levels.
Prior to Communication Manager release 2.1, you administer R2-Multifrequency Coded
(MFC) signaling parameters per system. With Communication Manager release 2.1 and
higher, you administer R2-MFC signaling parameters per trunk group.
R2-Multifrequency Coded signaling trunk groups use one of 8 sets of MFC signaling
parameters according to the MFC signaling code administered for that trunk group.
You can administer tone generation characteristics and administrator-entered customizations
per location. You can administer the server so that, when a telephone or trunk needs to play a
Communication Manager ECS-generated tone, the software plays a tone into the call using
the tone characteristics of the location of the listening endpoint, or of another endpoint on the
call.
Provides call retention, forced disconnect, intrusion, mode-of-release control, and re-ring to
switches on public networks. Different countries frequently refer to these capabilities by
different names.
World class tone detection enables Avaya Communication Manager to identify and handle
different types of call progress tones, depending on the system administration. You can use
the tone detector and identification to display on data terminal dialing and to decide when to
send digits on trunk calls through abbreviated dialing, ARS, AAR, and data terminal dialing.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Message Integration Features
Message Integration
Audible message
waiting
Avaya Interactive
Response
Centralized voice mail
through mode code
integration
Dual DCP I-channels
Enhanced feature
integrations for Avaya
Modular Messaging
Embedded AUDIX
Audible message waiting places a stutter at the beginning of the dial tone when a telephone
user picks up the telephone. The stutter dial tone indicates that the user has a message
waiting. This feature is particularly useful for visually impaired people who may not be able to
see a message light. It is often used with telephones that have no message waiting lights.
Audible message waiting may not be available in countries that restrict the characteristics of
dial tones provided to users.
The Avaya Interactive Response (IR) -formerly known as INTUITY Conversant® -voice
information system is an interactive voice-response system that automates telephone
transactions from simple tasks, like routing to the right department, to complex tasks, such as
registering college students or providing bank balances. It communicates with customers in
natural-sounding, digitally recorded speech, and performs 24-hours a day without the services
of an operator.
The system can handle single or multiple voice-response applications simultaneously, and
can serve up to 48 callers at once. It can operate by itself to dispense information or collect
data, or it can work with a host computer to access a large database such as bank account
records. With its speech-recognition capability, even rotary telephone users can have access
to sophisticated telephone-based services. Advanced telephone features provide intelligent
call-transfer capabilities and allow you to use the system in your existing telephone
environment.
The centralized voice mail feature eliminates the need for a voice mail system at each of the
sites in a network. It does so by allowing a network running Avaya Communication Manager
to use a single INTUITY AUDIX voice messaging system as a centralized voice mail system
that serves the whole network. The INTUITY AUDIX system can also serve as a centralized
voice mail system within a hybrid network of Communication Manager, DEFINITY BCS, and
Merlin Legend/Magix switches.
This feature supports the use of dual DCP I-channels for AUDIX networking. In this case,
networking refers to the ability to send data files between AUDIX systems, not to
communications with the switch.
This enhancement implements QSIG and IP integration for One-Step Recording, supporting
integration with Avaya Modular Messaging.
A new button type, audix-rec, is added to the Station screen for this feature. When
administered, the button requires the user’s Audix hunt group extension number along with it.
The new button type is not yet available on attendant consoles.
While many voice messaging systems require separate equipment and connections, the
embedded AUDIX system easily installs directly into your cabinet to support advanced voice
messaging capabilities without the need for an adjunct processor. Each embedded AUDIX
system supports up to 2000 mailboxes and stores up to 100 hours of recorded messages.
Special voice-processing features include voice mail, call answering, outcalling, multi-level
automated attendant, and bulletin board.
•
Shared extensions provide personal mailboxes for each person sharing a telephone.
•
Multiple personal greetings allows you to prepare a pool of up to nine personal
greetings to save time and provide more personal customer service. Separate
messages can indicate that you are on the telephone, away from the desk, on
vacation, etc. You can assign different messages to internal, external, or after-hours
calls.
•
Priority messaging places important messages ahead of others. Internal and external
callers can mark the message as priority.
•
Outcalling automatically dials a prearranged telephone number or pager when you
have messages in your voice mailbox.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Message Integration Features
Message Integration
•
Priority outcalling automatically dials a prearranged telephone number or pager when
you have priority messages in your voice mailbox.
•
Broadcasting allows you to send a single message to multiple recipients or to all
users on the system.
•
System broadcast allows you to send broadcast messages as regular voice
messages, or as messages that recipients hear as they log in.
•
AUDIX directory allows you to look up the extension number of any other user by
entering their name on the telephone keypad.
•
Personal directory allows you to create a list of nicknames for quick access to
telephone numbers.
•
Call answering for nonresident subscribers provides voice mailboxes for users who
do not have an extension number on the system.
•
Full mailbox answer mode informs callers whenever messages cannot be left
because there is no room in a subscriber mailbox.
•
Name record by subscriber lets you record your own name on the system.
•
Automatic message scan can play all new messages in part or in their entirety
without requiring you to press additional buttons, which is particularly useful when
you are getting messages from your mobile telephone.
•
Sending restrictions by community enables you to limit the communities of callers
who can communicate using AUDIX voice messaging.
•
Group lists allows you to create mailing lists of up to 250 people to use for
broadcasting messages.
•
Message forwarding allows you to forward messages with or without attached
comments.
•
Name addressing allows you to address messages by name if you do not know the
extension.
•
Private messaging is a special coding feature that prevents recipients from
forwarding messages.
•
Leave word calling allows you to press a button on your telephone in order to leave a
standard "call me" message on any extension.
•
Online help provides you with instant access to voiced instructions at any time when
you are using the system.
•
Multiple language support allows you to install up to nine languages on the system,
from a superset of 30 available languages.
•
INTUITY AUDIX
Enhanced message handling gives you the flexibility for handling messages. Two of
these features are optional advance/rewind that lets you advance through and
rewind individual messages, and undelete messages that lets you retrieve any
messages that you may have accidentally deleted.
INTUITY messaging solutions essentially offers the same user features as the embedded
AUDIX system, plus the following features:
•
Fax messaging allows you to handle faxes as easily as you handle voice mail. You
can send, receive, store, scan, delete, skip, or forward faxes. This feature is fully
integrated with voice messaging, so you can attach faxes to voice messages, for
example. You can also create special mailboxes for each of your fax machines.
These mailboxes accept fax telephone calls when the fax machine is busy and then
deliver the fax to the fax machine when the fax machine is available.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Message Integration Features
Message Integration
Avaya IA770 INTUITY
AUDIX messaging
application
•
Turn off AUDIX call answering allows you to turn off call answering in order to
conserve system resources. You can create a message that tells callers they cannot
leave a message, giving them another number to call, for example.
•
Pre-addressing allows you to address a message before recording it.
•
Integrated messaging allows you access and manage incoming voice, fax, and email messages and file attachments from your personal computer or your telephone.
A voice message will thus appear in your e-mail mailbox, for example, and vice
versa. You can also set options to have just the message headers appear in the
alternate mailbox. You can also create a voice or fax message by telephone and
send it to an e-mail recipient.
•
Text-to-speech allows you listen to a voice rendering of text messages sent from a
supported e-mail system and/or INTUITY message manager.
•
Print text allows you to print messages sent from a supported e-mail system and/or
INTUITY message manager.
•
Enhanced addressing allows you to send a message to up to 1500 recipients.
•
Transfer restrictions allow you to control toll fraud by restricting transfers going
through the voice messaging system.
•
Internet messaging allows you to exchange messages (voice and text) with any email address via the World Wide Web.
•
Avaya voice director allows you to address messages via spoken name, in addition
to using touchtones to enter extensions or names. It also supports transferring to
AUDIX subscribers, including those in other locations, by speaking a name.
• International availability.
The IA770 application enhances communications and information exchange within
enterprises, helping customers be more successful with call answering and messaging. The
IA770 application enables customers to see messages on their PCs, add a voice mail
component to an e-mail, and listen to e-mail using voice mail.
IA770 uses the Linux operating system, making it consistent with the operating system of the
G700 and G350 Media Gateways. The distributed architecture is designed for reliability and
survivability and is centrally managed for simplicity, efficiency and quick response to help
ensure business recovery.
The IA770 application consists of license file-activated software residing on the S8300 Media
Server, and a small card that can be installed and upgraded in the field.
The IA770 application includes INTUITY Message Manager. While the system provides textto-speech capability in U.S. English only, there is no additional charge for initial
implementation of any of the 35 available languages for prompts.
IA770 supports INTUITY digital (TCP/IP) networking protocol. More extensive networking can
be provided with the Avaya Interchange.
Using the Web interface, the administrator can perform a system backup and restore of all
administered data -announcements, recorded names, greetings -and approximately 50 hours
of messages over the local area network (LAN). The screens are easier to understand and
more intuitive, which should cut installation time and lessen the need for training and
experience. The IA770 system uses smart defaults rather than requiring every field to be
addressed.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Message Integration Features
Message Integration
S8100 Media Server
embedded INTUITY
AUDIX
This application provides voice, fax, and text messaging, along with text-to-speech and
message manager functionality in a single processor mezzanine board on the S8100 Media
Server. Note: Communication Manager release 3.0 is not available on the S8100 Media
Server. Included are Avaya Directory Enabled Management (DEM) and Fax Extended Dialing
(FED).
•
ADEM provides real time directory-based access to Communication Manager and
INTUITY AUDIX.
•
AUDIX one-step
recording
INTUITY call
accounting system
INTUITY lodging
INTUITY lodging call
accounting system
Leave Word Calling
FED allows the customer to specify restrictions on the destination numbers, as well
as eliminate the need to administer fax number ranges as remote AMIS networking
machines. Additionally, FED addresses the entry of international destination
numbers by allowing up to 23 digits for fax endpoints.
The INTUITY AUDIX mezzanine card also provides the necessary DSP resources for
messaging. This hardware eliminates the need for the INTUITY Map 5P adjunct, usually
required for this functionality.
Users can record conversations by pressing a single button. This feature uses AUDIX as the
recording device. This feature is not available with INTUITY AUDIX through Mode Codes or
remote AUDIX. Note: It it important that anyone who wants to activate this feature should
study and understand your local laws regarding the recording of calls before activating this
feature.
If you are using any of the INTUITY voice messaging products, the INTUITY call accounting
system is probably the best call-accounting solution for you. The system works exclusively
with INTUITY products, which reside on MAP/40 or MAP/100 computers. Offering many of
same features as the call accounting system for Windows, the system also serves to help
integrate your INTUITY applications.
INTUITY lodging is a messaging system designed especially for lodging establishments such
as hotels or other lodging providers such as hospitals or colleges. The system supplies guests
with electronic mailboxes that store voice or fax messages. INTUITY lodging serves as a
private answering machine for each extension.
Hotel guests can leave messages for each other without going through the attendant. For
incoming calls, an attendant transfers the call to the appropriate room. If the guest does not
answer the call or if the line is busy, the call is automatically transferred to the guest voice
mailbox, where the caller can leave a voice message. A message-waiting indicator on the
guest telephone notifies the guest that the voice mailbox contains messages. Guests are
assigned a password for accessing messages remotely. They can retrieve and save
messages from any telephone, on or off premises.
The INTUITY lodging call accounting package (an integrated offering from Homisco) takes
call records supplied by the system, puts the records into a standard bill format, and sends
the billing information to the property management system. When guests check out, their long
distance calling charges are printed automatically on their bill. This gives you better control
over telephone usage revenue.
Leave Word Calling (LWC) allows internal system users to leave a short preprogammed
message for other internal users. The preprogammed message usually is the word "call," the
caller name, extension, and the time of the call. When the message is stored, the message
lamp on the called telephone automatically lights.
Communication Manager can now handle up to 12,000 Leave Word Calling (LWC) messages.
LWC messages can be retrieved using a telephone display, voice message retrieval, or
AUDIX. Messages may be retrieved in English, French, Italian, Spanish, or a user-defined
language.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Message Integration Features
Message Integration
Leave Word Calling QSIG/DCS
Manual message
waiting
Message demand print
Message retrieval
Display retrieval
Speak-to-me
Message Sequence
Tracer enhancements
Mode code interface
Octel integration
The Leave Word Calling (LWC) feature is extended to enterprise networks with QSIG as the
private network protocol, as well as those with DCS.
For enterprise networks that are mixed or in transition from DCS to QSIG, interworking of the
LWC feature between the protocols can be provided. LWC also works within a single nonnetworked switch. Note: A DCS+ signaling group is needed, but can only be used in networks
with 4-digit or 5-digit dial plans.
This feature allows multi-appearance telephone users to light the status lamp associated with
the manual message waiting button at another multi-appearance telephone. They do this by
simply pressing a button on their own telephone. This feature can be administered only to
pairs of telephones such as a secretary and an executive. The secretary might press the
button to signal to the executive that a call needs answering or someone has arrived for an
appointment. The executive might use the button to indicate that he or she should not be
disturbed.
Message demand print allows you to print your undelivered messages without calling the
message center.
With the message waiting lamp on their telephones, employees always know when they have
messages. Messages can be retrieved in a variety of ways. These message retrieval options
can be assigned to individual users.
Users having digital telephones with displays or a personal computer integrated with a
telephone can display messages.
Using any touch-tone telephone, employees can dial speak-to-me and hear a synthesized
voice read their messages over the telephone.
In the past, it had been difficult to trace messages through the Message Sequence Tracer
(MST) tool pertaining to a particular socket because there was no tag in each message
distinguishing it from other sockets.
New message formats for outgoing and incoming data now include the socket
number/identifier. These new formats use new Type identifiers of 05 and 06. A pair of new
formats 07 and 08 have also been created for outgoing and incoming socket control
messages on the PROCR ip-interface.
By creating new format types for these new formats, the task of decoding these messages is
easier.
The following enhancement was made to the Message Sequence Tracer (MST):
•
Signaling messages between Communication Manager and the TN799 CLAN can
now be traced for better diagnostics during network outages. -Add processor TN799
CLAN socket information to the MST trace in order to help developers debug socket
problems.
•
Enhance MST to include the socket number in socket data.
• Add TN799 CLAN board ID to CLAN MST IP socket trace messages.
Communication Manager supports an analog mode code interface for communications with
INTUITY AUDIX and other voice mail systems using the same interface. This interface
employs DTMF tones, line signals, and feature access codes, and allows INTUITY AUDIX to
exchange data with Communication Manager without using a data link. Other adjunct vendors
can engineer their products to use this interface.
Communication Manager integrates with the entire line of Octel messaging systems including
the Octel 200/300 message server, and the Octel 250/350 message server.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Message Integration Features
Message Integration
QSIG/DCS voice mail
interworking
Multiple QSIG voice
mail hunt groups
Voice mail retrieval
button
Voice message
retrieval
QSIG/DCS voice mail interworking is an enhancement to the QSIG feature. It integrates DCS
and QSIG centralized voice mail using the DCS+/QSIG gateway. Switches labeled
DCS+/QSIG integrate multi-vendor PBXs into a single voice messaging system. QSIG/DCS
voice mail interworking works on G3r, G3si, and G3csi. It provides network flexibility, DCS
functionality without a dedicated T1.
Communication Manager provides for ten message center hunt groups to support QSIG
integrated messaging. This feature allows customers to spread users in a single
Communication Manager system over multiple messaging systems. This allows users to
move among Communication Manager systems while retaining their same voice mailbox.
Users do not lose voice messages.
This feature also enhances customer usability of Avaya messaging systems in the enterprise
by allowing not only for growth, but the ability to migrate end users on a single
Communication Manager system.
Avaya Communication Manager supports the voice mail retrieval feature as a fixed feature
button on the 2420 DCP and the 4602 telephone.
A field, "voice-mail Number: _______" appears on the Station screen for stations of type
2420 and 4602. The allowed values for this field are identical to the values allowed for an
autodial feature button number. The field is a fixed field allowing entry of up to 16 digits that
are auto-dialed to access the user’s voice mail system.
If the number field is blank, the voice mail retrieval button is treated like the "Transfer to Voice
Mail" button.
If the number field is not blank, the voice mail retrieval button is treated like an autodial button.
Voice message retrieval allows telephone users, remote access users, and attendants to
retrieve leave word calling and call coverage voice messages. You can use voice message
retrieval to retrieve your own messages or messages for another user. However, you can only
retrieve messages for another user:
•
Voice messaging and
call coverage
from a telephone or attendant console in the coverage path
• from an administered system-wide message retriever
if you are a remote-access user and you know the extension and associated security code
The system restricts unauthorized users from retrieving messages.
Often an AUDIX system is set up as the last point on a call-coverage path. A secretary or
colleague who answers a redirected call intended for you can also transfer the caller to your
AUDIX mailbox. The caller may prefer to leave voice-mail for you if the message is personal,
lengthy, or technical.
Many other options are available. For example, a caller can redirect a call from the AUDIX
system to an attendant. Or the caller can transfer to another extension instead of leaving a
message. You can even have the AUDIX automated attendant answer all calls to the
company and send calls to various extensions. In this case, callers are instructed to enter
keypad commands to direct the call.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Mobility Features
Mobility
IP telephones or IP Softphones allow you to access the features of Communication Manager from anywhere. With IP
telephones is that you can move the telephones around on a LAN just by unplugging and re-plugging. With IP softphones is
that you can load them on a laptop PC, and then connect them to the switch from almost anywhere.
Administration
This feature allows you to administer telephones that are not yet physically present on the
Without Hardware
system. This feature works the same as administration with hardware: when stations are
moved, user-activated features such as call forwarding and send all calls are preserved and
functional. This greatly facilitates the speed of setting up and making changes to the
telephones on the system.
Automatic Customer
Automatic Customer Telephone Rearrangement (ACTR) allows a telephone to be unplugged
Telephone
from one location and moved to a different location without additional switch administration.
Rearrangement
The switch automatically associates the extension to the new port.
ACTR works with the 2420 DCP telephone and the 6400 serialized telephones. The 6400
serialized telephone is stamped with the word "serialized" on the faceplate for easy
identification. The 6400 serialized telephone memory electronically stores its own part ID
(comcode) and serial number. ACTR uses the stored information and associates the
telephone with new port when the telephone is moved.
ACTR makes it easy to identify and move telephones.
Avaya Wireless
Avaya Wireless Telephone Solutions (AWTS) is fully integrated with Communication
Telephone Solutions
Manager, and allows a user full access to Communication Manager features from a mobile
telephone. Note: Avaya Wireless Telephone Solutions (AWTS) replaces the DEFINITY
Wireless Business System (DWBS).
Avaya Extension to
The Avaya Extension to Cellular feature provides the expansion of mobile services, including
Cellular
one-number availability, increased user capacities, flexibility across facilities and hardware,
more control over unauthorized usage, enhanced enable/disable capability, increased
serviceability, and support of IP trunk facilities.
Avaya Extension to Cellular and off-PBX stations (OPS) provides users with the capability to
have one administered telephone that supports Avaya Communication Manager features for
both an office telephone and one outside telephone. Extension to cellular/OPS allows users to
receive and place office calls anywhere, any time. People calling into an office telephone can
reach users even if they are not in the office. Users could receive the call on their cell
telephone, for example.
This added flexibility also allows them to use certain Communication Manager features from a
telephone that is outside the telephone network.
Previous versions of Extension to Cellular allowed for office calls to be extended to the cell
telephone of a user. Also, calls from the cell telephone would appear as if the call originated
from the user office telephone when calling another telephone on the same call server.
Certain features within Communication Manager are available from the cell telephone. These
features are still available.
In previous versions of Extension to Cellular, cell telephones had to be administered as
XMOBILE stations. This is no longer necessary with Communication Manager Release 2.0.
If you had administered Extension to Cellular in earlier releases of Communication Manager,
you do not have to change the administration to continue using Extension to Cellular features.
It still works. However, users would not have the full range of features that are now possible
with Extension to Cellular/OPS.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Mobility Features
Mobility
IP telephones or IP Softphones allow you to access the features of Communication Manager from anywhere. With IP
telephones is that you can move the telephones around on a LAN just by unplugging and re-plugging. With IP softphones is
that you can load them on a laptop PC, and then connect them to the switch from almost anywhere.
Off-PBX station
With Avaya Communication Manager Release 2.0, the off-PBX station (OPS) application type
is used to administer of a SIP telephone. OPS cannot be disabled using the Extension to
Cellular enable/disable feature button. Note: A 4602 SIP telephone must register with the SIP
proxy regardless of whether OPS is administered.
The Extension to Cellular/OPS application allows for many of the parameters used for the
original Extension to Cellular application to be ported onto one of several DCP and IP station
types. From a call processing perspective, Extension to Cellular/OPS is in fact dealing with a
multi-function telephone, whereas the previous Extension to Cellular implementation utilized
one or two XMOBILE stations that behaved like analog station types.
E911 ELIN for IP wired This feature automates the process of assigning an emergency location information number
extensions
(ELIN) through an IP subnetwork ("subnet") during a 911 call. The ELIN is then sent over
either CAMA or ISDN PRI trunks to the emergency services network when 911 is dialed. This
feature properly identifies locations of wired IP telephones that call an emergency number
from anywhere on a campus or location. Note: This feature depends upon the customer
having subnets that correspond to geographical areas.
This feature works for both types of IP endpoints:
•
E911 device location
for IP telephones
Personal Station
Access
Do not answer reason
code
Name/number
permanent display
H.323
• SIP
Without this feature, if these users dial 911, the emergency response personnel might go to
the wrong physical location. With this feature, the emergency response personnel can now go
to the correct physical location. In addition, emergency response personnel can now go to the
correct physical location if a 911 emergency call comes from a bridged call appearance.
Communication Manager works with an E911 Manager device from RedSky Technologies.
This third-party E911 Manager provides a flexible, complete, and automated E911
management system for customers who want to implement voice over IP (VoIP) telephony.
The E911 Manager from RedSky Technologies works with Communication Manager release
2.1 and beyond to keep the Automatic Location Information (ALI) record for each extension
correct. The E911 Manager also provides notification whenever someone moves an IP
endpoint to a new subnet.
The Personal Station Access (PSA) feature allows you to transfer your telephone station
preferences and permissions to any other compatible telephone. This includes the definition
of telephone buttons, abbreviated dial lists, and class of service, and class of restrictions
permissions.
PSA has several telecommuting applications. For example, several telecommuting employees
can share the same office on different days of the week. The employees can easily make the
shared telephone "theirs" for the day.
The Personal Station Access (PSA) feature uses Administration Without Hardware (AWOH),
a feature that allows the system administrator to assign a telephone without specifying a
physical port. For example, use "X" as the port. If a telephone is disassociated, it means that it
is not currently mapped to a particular physical telephone, such as a digital telephone. If a
caller dials into an extension that is currently disassociated, they are provided a message that
indicates "Don't answer" instead of "Busy".
When a person uses PSA to associate their extension with a station, a display appears on the
station indicating their name and extension number. This information is displayed until the
user disassociates their extension from the station using the PSA-associate feature access
code.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Mobility Features
Mobility
IP telephones or IP Softphones allow you to access the features of Communication Manager from anywhere. With IP
telephones is that you can move the telephones around on a LAN just by unplugging and re-plugging. With IP softphones is
that you can load them on a laptop PC, and then connect them to the switch from almost anywhere.
Enterprise Mobility User Enterprise Mobility User (EMU) is a software-only feature that gives you the ability to
associate the buttons and features of your primary telephone to a telephone of the same type
anywhere within your company enterprise.
Note: In this document, any telephone that is not the primary telephone is referred
to the visited telephone and any server that is not the home server of the primary
telephone is referred to as the visited server.
The following is a list of requirements that you need for the EMU feature:
•
QSIG must be the private networking protocol in the network of Communication
Manager systems.
•
Communication Manager Release 3.1 and later software must be running on the
home server and all visited servers.
•
All servers must be on a Linux platform. EMU is not supported on DEFINITY servers.
•
The visited telephone must be the same model type as the primary telephone to
enable a optimal transfer of the image of the primary telephone. If the visited
telephone is not the same model type, only the call appearance (call-appr) buttons
and the message waiting light are transferred.
•
EMU is only supported on self-designating terminals (terminals with button labels)
that are downloaded from the Communication Manager server.
• Uniform Dial Plan (UDP).
How Enterprise Mobility User works
On the dial pad of a visited telephone, a user enters the EMU activation feature access code
(FAC), the extension number of their primary telephone, and a security code. The visited
server sends the extension number, the security code, and the set type of the visited
telephone to the home server.
When the home server receives the information, the home server:
•
Checks the Class of Service (COS) for the primary telephone to see if it has PSA
permission.
•
Compares the security code with the security code on the Station screen for the
primary telephone.
•
Compares the station type of the visited telephone to the station type of the primary
telephone. If both the visited telephone and the primary telephone are of the same
type, the home server sends the applicable button appearances to the visited server.
If a previous registration exists on the primary telephone, the new registration is
accepted and the old registration deactivated.
If the registration is successful, the visited telephone assumes the primary telephone’s
extension number and some specific administered button types. The display on the primary
telephone shows Visited Registration Active: <Extension>. The <Extension> that displays is
the extension number of the visited telephone.
Note: The speed dialing list that is stored on the primary telephone and the station
logs are not downloaded to the visited telephone. EMU does not allow users to
associate permissions from the home telephone to the remote telephone.
©2006 Avaya Inc.
Page 40
Avaya Communication Manager Feature Overview
Mobility Features
Mobility
IP telephones or IP Softphones allow you to access the features of Communication Manager from anywhere. With IP
telephones is that you can move the telephones around on a LAN just by unplugging and re-plugging. With IP softphones is
that you can load them on a laptop PC, and then connect them to the switch from almost anywhere.
Terminal Translation
Communication Manager provides Terminal Translation Initialization (TTI), a feature that
Initialization
works with Administration Without Hardware (AWOH). TTI associates the terminal translation
data with a specific port location through the entry of a special feature-access code, a TTI
security code, and an extension number from a terminal that is connected to a wired (but
untranslated) jack.
X-station mobility
X-station mobility allows remote users to access switch features. That is, X-station mobility
allows certain OEM wireless telephones remoted over a PRI trunk interface to be controlled
by Communication Manager as if the telephones were directly connected to the switch.
The telephones are administered to be of the type XMOBILE and have additional
administration information on the Station screen that assigns the capabilities of a remote
station to the associated PRI trunk group. The wireless telephones thus have access to such
features as call-associated display, bridging, message waiting, call redirection, and so forth.
X-station mobility is currently used for non-cellular wireless offers (DECT and PHS) in EMEA
and APAC regions, and the Extension to Cellular offer globally.
©2006 Avaya Inc.
Page 41
Avaya Communication Manager Feature Overview
Port Network and Gateway Connectivity Features
Port Network And Gateway Connectivity
Asynchronous
Transfer Mode
ATM WAN Spare
Processor Manager
Port Network
Connectivity
Port Network
Connectivity over
WAN
WAN Spare Processor
The Asynchronous Transfer Mode (ATM) switch is a replacement option for the CSS or the
direct-connect switch. Several Avaya ATM switch types can provide Communication Manager
port network connectivity. Non-Avaya ATM switches that comply with the ATM standards set
by the European Union can also provide Communication Manager port network connectivity.
ATM WAN Spare Processor (WSP) Manager can be a key part of your emergency restoration
and business continuity planning. This application enables users to download translations
from a main server running Communication Manager, and simultaneously upload those
translations to multiple (up to 15) ATM WAN Spare Processors (WSPs) over a LAN
connectivity. This can be done according to a schedule specified by the administrator.You can
schedule translations to run once now, or for a specified time and date in the future. You can
also schedule regular daily or weekly updates.
The module also provides the ability to schedule regular daily or weekly updates of the
Communication Manager translations. The ATM WAN Spare Processor Manager provides the
current status of the main server running Communication Manager and any defined WSP
devices in the network. A complete history log is created listing each of the switches, and the
time and the resulting message from the scheduled action. On-line help is embedded into the
module for ease of use.
ATM Port Network Connectivity (ATM-PNC) provides an alternative to the Center Stage
Switch (CSS) configurations for connecting the Processor Port Network (PPN) to one or more
Expansion Port Networks (EPN). ATM-PNC replaces the CSS in a DEFINITY R8r and later
network with an ATM switch or network. ATM-PNC is available with all three Communication
Manager reliability options -standard, high, and critical. In addition, it offers ATM-PNC
duplication.
ATM-PNC integrates delivery of voice, video, and data via ATM over a converged large
bandwidth network, providing reduced infrastructure cost and improved network
manageability. ATM-PNC uses standards-based open interfaces that can be provisioned with
either new or existing systems running Communication Manager.
ATM-PNC over a public Wide Area Network (WAN) represents an environment where the
customer uses a service provider’s ATM network between privately-owned ATM switches.
The customer does not control the ATM switches in the network, including traffic policing
policies and product quality.
Using a public WAN, Permanent Virtual Paths (PVP) may be set up between customer-owned
ATM switches similar to the dedicated circuits in a private WAN. However, ATM cell
processing occurs in a public WAN so the customer is dependent on ATM switches owned
and managed by the service provider.
Switched Virtual Circuits (SVC) use the ATM protocol to transmit voice-like applications over
ATM networks. The advantage of the SVC solution is that Communication Manager can
dynamically signal the ATM network to provide more bandwidth as needed to handle peaks in
the call traffic. If the ATM network cannot handle the additional traffic, calls will be denied.
An ATM WAN Spare Processor (WSP) provides a disaster recovery option for a media
gateway G3r expansion port networks deployed over an ATM WAN.
An ATM WSP acts as a PPN in the event of a catastrophic failure in the network. The ATM
WSP continually monitors a path to the PPN to determine if it is on-line and reachable. The
WSP functions as a PPN if the main PPN is not functional or is not communicating to one or
more of the other EPNs. From one to 15 ATM WSPs can be placed in a Communication
Manager ATM port network configuration to provide a backup arrangement of PPNs, thus
maintaining the availability of the Communication Manager features and functions. Note: ATM
WSPs cannot be used with a conventional CSS.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Port Network and Gateway Connectivity Features
Port Network And Gateway Connectivity
Banner displayed to
warn of reset
Block circuit pack
installation if wrong
suffix
When a new license file is loaded which changes the value of FEAT_ESS from that of the
previous license files, a “reset sys 4“ is required in order for the change to take effect.
If the “reset sys 4“ is not done, a banner is displayed on the initial SAT screen warning the
user that a reset is required.
In an Enterprise Survivable Server (ESS) system:
•
When a TN2305 or TN2306 Asynchronous Transfer Mode (ATM) Expansion
Interface (EI) suffix “A” circuit pack is inserted in a position where a suffix “B” circuit
pack is required, the circuit pack insertion is prevented and an alarm (MINOR
ON_BOARD) is raised.
•
H.248 media gateway
control
Inter-Gateway
Alternate Routing
When a TN750 EI circuit pack, with a suffix other than “D,” is inserted in a position
where a suffix “D” circuit pack is required, the circuit pack insertion is prevented and
an alarm (MINOR ON_BOARD) is raised.
Internet Protocol
Communication Manager uses standards based H.248 to perform call control to Avaya media
gateways such as the G700. H.248 defines a framework of call control signaling between the
intelligent media servers and multiple "unintelligent" media gateways.
For single-server systems that use the IP-WAN to connect bearer between port networks or
media gateways, Inter-Gateway Alternate Routing (IGAR) provides a means of alternately
using the public switched telephone network (PSTN) when the IP-WAN is incapable of
carrying the bearer connection. IGAR may request that bearer connections be provided by the
PSTN under the following conditions:
•
The number of calls allocated or bandwidth allocated via Call Admission ControlBandwidth Limits (CAC-BL) has been reached
•
VoIP RTP resource exhaustion in a MG/PN is encountered
•
A codec set is not specified between a network region pair
•
Forced redirection between a pair of network regions is configured
•
IGAR takes advantage of existing public and private network facilities provisioned in
a network region. Most trunks in use today can be used for IGAR. Examples of the
better trunk facilities for use by IGAR are:
•
Public or Private ISDN PRI/BRI
• R2MFC
IGAR provides enhanced Quality of Service (QoS) to large distributed single-server
configurations.
Network Region
Wizard
IP Port Network
Connectivity
Link Recovery
For large distributed single-server systems that have multiple network regions, the Network
Region Wizard (NRW) simplifies and expedites the provisioning of multiple IP network
regions, including Call Admission Control using Bandwidth Limits (CAC-BL) and InterGateway Alternate Routing (IGAR).
Communication Manager allows Control Channel Message Set (CCMS) messages to be
packetized over IP LAN and WAN connections to control multiple port networks.
IP calls must have an H.248 link between the Avaya G700 Media Gateway and the call
controller. The H.248 link between an Avaya server running Communication Manager and the
Avaya Media Gateway provides the signaling protocol for:
•
Call setup
•
Call control (user actions such as Hold, Conference, or Transfer)
•
Call tear-down
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Port Network and Gateway Connectivity Features
Port Network And Gateway Connectivity
Separation of Bearer and
Signaling
Enhanced TN2602AP
circuit pack
If the link fails for any reason, the Link Recovery feature preserves any existing calls and
attempts to re-establish the original link. If the gateway cannot reconnect to the original
server, then Link Recovery automatically attempts to connect with alternate TN799DP (CLAN)
circuit packs within the original server configuration or to a Local Spare Processor (LSP).
Link Recovery does not attempt to recover or overcome any network failure that created the
link outage. Link Recovery also does not diagnose or repair the network failure that caused
the link outage.
Since there is no communication possible between the Media Gateway and call controller
during a link outage, button depressions are not recognized, feature access does not work,
and neither does any other type of call handling. In essence, the system is unresponsive to
any stimuli until the H.248 link is restored. This might be the only indication that a Link
Recovery is in process.
The Separation of Bearer and Signaling (SBS) feature provides a low cost virtual private
network with high voice quality for customers who cannot afford private leased lines. SBS
provides a DCS+ VPN replacement for those customers needing Dial Plan Expansion (DPE)
functionality. Note: DCS does not work with six-digit or seven-digit dial plans. Although QSIG
does work with six-digit and seven-digit dial plans, QSIG does not work over VPNs.
The SBS feature supports:
•
QSIG private networking signaling over a low cost IP network
•
Voice (bearer) calls over public switched network
• Association between QSIG feature signaling information and each voice call
You must always use AAR/ARS/UDP to originate an SBS call. You cannot use a Trunk
Access Code / Dial Access Code to originate an SBS call.
Communication Manager release 3.1 includes enhancements to the TN2602AP circuit pack,
described in the following paragraphs.
Note: The TN2602AP IP Media Resource 320 is not supported in CMC1 and G600
Media Gateways. For more information about the TN2602AP circuit pack, see the
Hardware Description and Reference for Avaya Communication Manager, 555-245207.
Bearer signal duplication
The capabilities of the TN2602AP circuit pack have been expanded to provide duplicated
bearer support. This enables customers to administer IP-PNC with critical bearer reliability. A
port network continues to support a maximum of two TN2602AP circuit packs, but they can
now be administered for duplication. This capability is in addition to the previously-offered
load balanced support (see Load balancing).
Two TN2602AP circuit packs may be installed in a single port network for bearer signal
duplication. In this configuration, one TN2602AP is an active IP media processor and one is a
standby IP media processor. If the active media processor, or connections to it, fail, active
connections failover to the standby media processor and remain active. This duplication
prevents active calls in progress from being dropped in case of failure.
Duplicated TN2602AP circuit packs operate in an Active-Standby mode. State of health
parameters exist between the two boards to determine when it is appropriate to interchange
duplicated TN2602AP circuit packs. It is also possible to manually invoke an interchange
using a software command.
For bearer duplication, both TN2602AP circuit packs must be Hardware Version 2, and must
have firmware version 211 or higher.
Note: The 4606, 4612, and 4624 telephones do not support the bearer duplication
feature of the TN2602AP circuit pack. If these telephones are used while an
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Port Network and Gateway Connectivity Features
Port Network And Gateway Connectivity
Load balancing
Reduced channels with
duplicated TN2602AP
circuit packs
interchange from active to standby media processor is in process, calls might be
dropped.
Important: If you change from load balanced to duplicated TN2602s, existing calls
retain the real IP address on the TN2602AP circuit pack. New calls are associated
with the virtual IP address of the TN2602AP circuit pack. If an interchange occurs
during this time, existing calls that are associated with the real IP address will drop.
Up to two TN2602AP circuit packs can be installed in a single port network for load balancing
or duplication. When in a load balanced mode, calls are distributed evenly among the two
TN2602 circuit packs.
The TN2602AP circuit pack is also compatible with, and can share load balancing with, the
TN2302 and the TN802B IP Media Processor circuit packs. Actual capacity may be affected
by a variety of factors, including the capacity of the circuit pack being used, the codec used
for a call, and fax support.
Note: If duplicated TN2602 circuit packs are combined with a TN2302 or TN802,
Communication Manager uses the active, duplicated TN2602 to capacity before
using another media processor circuit pack. Also, when media processor circuit
packs in the same port network are in different network regions, load balancing
does not apply.
If a pair of TN2602AP circuit packs, previously used for load balancing, are re-administered to
be used for bearer duplication, only the voice channels of the active circuit pack can be used.
For example,
•
If you have two TN2602 AP circuit packs in a load balancing configuration, each with
80 voice channels, and you re-administer the circuit packs to be in bearer
duplication mode, you have 80, not 160, channels available.
•
If you have two TN2602 AP circuit packs in a load balancing configuration, each with
320 voice channels, and you re-administer the circuit packs to be in bearer
duplication mode, you will have 320, not the maximum 484, channels available.
•
Increased trunk
members for IP signaling
groups
Incremental filesyncs
More BRI Trunk circuit
packs
More than nine static
routes allowed
Music on hold played
from nearest source
H.248 and H.323
registration
When two TN2602AP circuit packs, each with 320 voice channels, are used for load
balancing within a port network, the total number of voice channels available is 484,
not 640. The reason is that 484 is the maximum number of time slots available for
connections within a port network.
The number of H.323 trunk members in a single signaling group that are supported on the
Trunk Groups screen is increased from 31 to 255. Users also have the option to administer
each trunk group member individually or automatically.
The system now supports two difference sets for incremental filesyncs, one for LSPs and one
for ESSs.
An S8700, S8710, S8500, or S8300 Media Server running Communication Manager can now
have up to 250 TN2185 (BRI Trunk) circuit packs.
A radio button is added to the Set Static Routes section of the configure server web pages
that allows more than nine static routes.
An IP telephone in one network region (the “calling party”) calls another IP telephone in
another network region (the “called party”). If the calling party places the called party on hold,
the called party hears music from the nearest music source. Assuming that the gateways in
both network regions had music installed, music would come from the called party’s gateway.
The system uses the PE interface on an LSP to register H.248 gateways and H.323
endpoints. Starting with Communication Manager Release 3.1, the use of the PE interface to
register H.248 gateways and H.323 endpoints has been expanded to include the simplex
main server.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Port Network and Gateway Connectivity Features
Port Network And Gateway Connectivity
Inter-Gateway Alternate
Routing (IGAR) calls
over Inter-Gateway
Connections
The output of the command display internal-data s-tab now includes information about IGAR
(Inter-Gateway Alternate Routing) calls using shared connection (“dumbbell”) topology over
Inter-Gateway Connections (IGCs). An IGAR call can use up to 120 IGCs.
A new field igccount' displays the total quantity of dumbbell IGCs used by an IGAR call.
Dumbbell information for those IGCs appears on additional pages of the output.
The information listed for each PSTN IGC includes:
•
Fabric Type (PSTN)
•
Master Trunk Port ID
• Slave Trunk Port ID
The information listed for each IP IGC includes:
•
Fabric Type (IP)
•
Master Ephemeral IP Address (on a port network or gateway)
•
Master Ephemeral Port
•
Slave Ephemeral IP Address (on a port network or gateway)
•
Slave Ephemeral Port
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Trunk Connectivity Features
Trunk Connectivity
Asynchronous Transfer
Mode
Circuit Emulation
Service
CMS measurement of
ATM
Gateway trunk
preference selection
Local ringback
administration
Parameterized data for
NSF
Prepend '+' to calling
number
DS1 trunk service
Echo cancellation -with
UDS1 circuit pack
E1
See Asynchronous Transfer Mode.
ATM-circuit emulation service (ATM-CES) lets Communication Manager emulate ISDN-PRI
trunks on an ATM facility. These virtual trunks can serve as integrated access, tandem, or tie
trunks.
ATM-CES trunk emulation maximizes port network capacities by consolidating trunking. For
example, the CES interface can define up to eight virtual circuits for tie-line connectivity,
consolidating onto one circuit card network connectivity that usually requires multiple circuit
packs.
See CMS measurement of ATM.
When trunks from more than one Media Gateway (G350, G700, or G250) are in the same
trunk group, Communication Manager new “prefers” trunks on the same Media Gateway as
the originator.
A new field is added to the Trunk Group screen, allowing the administrator to set if local
ringback tone should be sent to a caller.
If the Apply Local Ringback? field is set to y, and the system does not receive a PI_IBI in
ALERT, then the system sends a local ringback tone to the caller. The local ringback tone is
removed when the system receives a connect, and the channel will cut through.
The isdn network-facilities screen now provides a new column, administrable for user-entered
Network Specific Facility (NSF) names. The value in this column indicates whether the NSF
handles parameterized data. The default value is n, but you can change it to y.
When this value is set to y for outwats-bnd or any user-administered NSF name, you can see
the Parm column on the route-pattern screen. The value in this column provides information
for handling the parameterized data. For instance, if the NSF is SCOCS, it defines the class of
service requested for the parameterized data. It is blank by default, but you can give it any
numeric value up to 5 digits.
The SIP Trunk Group screen now provides the field Prepend + To Calling Number. The
default setting is n. If you set the field to y, the character + is added at the beginning of the
calling number for that trunk group.
Circuit switched
Bit-oriented signaling that multiplexes 24 channels into a single 1.544-Mbps stream. DS1 can
be used for voice or voice-grade data and for data-transmission protocols. E1 trunk service is
bit-oriented signaling that multiplexes 32 channels into a single 2.048-Mbps stream. Both DS1
and E1 provide a digital interface for trunk groups. Digital Service 1 (DS1) trunks can be used
to provide T1 or ISDN Primary Rate Interface (PRI) service.
The universal DS-1 (UDS1) circuit pack (TN464GP/TN2464BP) available for all
Communication Manager platforms has echo cancellation circuitry. The echo cancellation
capability of the circuit pack is intended only for channels supporting voice communication. It
is not desirable to provide echo cancellation over channels supporting data communication.
The TN464GP/TN2464BP is intended for Communication Manager customers who are likely
to encounter echo over circuits connected to the public network. The occurrence of echo is
likely if Communication Manager is configured for complex services such as ATM or IP. In
addition, echo is likely to occur if Communication Manager interfaces to local service
providers who do not routinely install echo cancellation equipment in all their circuits.
Communication Manager also supports E1 connections. T1/E1 access and conversion allows
simultaneous connection to both T1 (1.544 Mbps) and E1 (2.048 Mbps) facilities (using
separate circuit packs).
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Trunk Connectivity Features
Trunk Connectivity
T1
Separate licensing for
TDM stations and TDM
trunks
H.323 trunk
IP loss groups
When planning your networking requirements, one of the options you should consider is
multiplexing over digital services 1 (DS1) facilities.
Prior to release 2.0, Communication Manager was sold by licensed ports that included
stations and trunks. The system displayed the total of licensed ports in the Maximum Ports
field on the Optional Features screen.
As of release 2.0 of Communication Manager, Avaya sells licenses for stations, but not
trunks. Currently, the Maximum Ports field on the Optional Features screen is used for
licensing ports, which include both trunks and stations.
With Communication Manager release 3.0, a separate field, Maximum Stations, is created
on the Optional Features screen to track station licenses only. This helps customers easily
identify the number of station licenses on the system.
Internet Protocol
A TN802B in MedPro mode or a TN2302AP IP interface enables H.323 trunk service using IP
connectivity between two systems running Communication Manager. The H.323 trunk groups
can be configured as system-specific tie trunks, generic tie trunks, or direct-inward-dial (DID)
public trunks. In addition, the H.323 trunks support ISDN features such as QSIG and BSR.
A primary reason to accomplish a loss plan for voice communication systems is the desire to
have the received speech and tone loudness at a comfortable listening level. This should be
accomplished so that users can listen to each other without being concerned who or where
the remote party is, or what kind of telephone equipment each may be using.
A connection with an end-to-end loss (called an Overall Loudness Rating) of 10 dB -which
approximates a normal conversation between a talker and listener spaced one meter apart provides a high degree of satisfaction for the majority of users. Therefore, voice
communication standards for end-to-end loss are based on this number.
Communication Manager has now defined two additional loss groups for IP telephony. The
purpose of these two loss groups is to set speech and tone loudness separately for IP
connections. These loss groups use country-specific gateway loss plans.
The two IP loss groups are:
•
IP trunks
Loss Group 18: IPtrunk -loss group for IP trunks (IP Carrier Medium)
• Loss group 19: IPphone -loss group for IP terminals (IP ports)
On an upgrade, if the default for an IP station loss plan is 2, and the IP trunk loss plan is 13,
Communication Manager changes the defaults to 19 and 18 respectively.
IP trunk groups may be defined as virtual private network tie lines between systems or ITS-E
servers running Communication Manager. Each IP trunk circuit pack provides a basic 12-port
package that can be expanded up to a total of 30 ports. The number of ports that are defined
will correspond to the total number of simultaneous calls transmitted over the IP trunk
interface.
The benefits of IP trunk include a reduction in long distance voice and fax expenses,
facilitating global communications, providing a full function network with data and voice
convergence and optimizing networks by using the available network resources.
IP trunking is a good choice for basic, corporate voice and fax communications, where cost is
a major concern. IP trunk calls travel over a company intranet rather than the public telephone
network. So, for the most common types of internal corporate communications, IP trunks offer
considerable savings.
IP trunking is usually not a good choice for applications where calls have to be routed to
multiple destinations (as in most conferencing applications) or to a voice messaging system.
IP trunk calls are compressed to save network bandwidth. Repeated compression and
decompression results in a loss of data at each stage and degrades the final quality of the
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Trunk Connectivity Features
Trunk Connectivity
Session Initiation
Protocol
SIP trunks
signal.
The maximum number of compression cycles acceptable on a call is three, and three
compression cycles can compromise voice quality. Normal corporate voice or fax calls
typically go through fewer than three compression cycles. However, multipoint conference
calls and most voice messaging systems add too many compression cycles for acceptable
quality.
Session Initiation Protocol (SIP) is a signalling protocol used for establishing sessions in an IP
network.
SIP has a separate set of feature documentation available at http://www.avaya.com/support
SIP trunking functionality allows a Linux server to function as a POTS gateway between
traditional legacy endpoints (stations and trunks) and SIP endpoints. It also provides SIP to
SIP routing. In the routing scenario, the server supports call routing similar to what a SIP
proxy would provide.
SIP links can be secured using TLS to encrypt signaling, and use Digest Authentication to
perform validation. When using TLS, the Media Encryption feature is also available to encrypt
audio channels.
SIP trunking functionality:
•
Provides access to less expensive local and long distance telephone services, plus
other hosted services from SIP service providers
•
Provides presence and availability information to members of the enterprise and
authorized consumers outside the enterprise, including other enterprises and service
providers
•
Auxiliary trunks
Advanced Private Line
Termination
Central Office
Central Office support
on G700 Media
Gateway -Russia
Facilitates SIP-enabled converged communications applications within the
enterprise, such as the Seamless Service Experience.
Allowing encryption of signaling and audio channel provides the customer with the option to
provide a secure communications infrastructure.
Auxiliary trunks connect devices in auxiliary cabinets with Communication Manager. Some of
the features that are supported with this type of trunk are recorded announcements,
telephone dictation service, malicious call trace, and loudspeaker paging.
Provides access to and termination from CO (Central Office)-based private networks; namely,
Common Control Switching Arrangements (CCSA) and Enhanced Private Switched
Communications Service (EPSCS).
APLT trunks are physically the same as those used for analog tie trunks, where the trunk
signaling is compatible with EPSCS and CCSA network switches. The outgoing APLT trunk
repeats any number of digits to the private network as dialed. APLT trunks can tandem
through the PBX from EPSCS network only; CCSA networks require an Attendant to
complete the call.
Central Office (CO) trunks connect Communication Manager to the local central office for
incoming and outgoing calls.
See Central Office support on G700 Media Gateway.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Trunk Connectivity Features
Trunk Connectivity
Digital multiplexed
interface
Bit-oriented signalling
Message-oriented
signalling
Direct Inward Dialing
Direct Inward/Outward
Dialing
E&M signaling continuous and pulsed
E911 CAMA trunk
group
Foreign Exchange
ISDN trunks
Automatic Termination
Endpoint Identifier
Call-by-call service
selection
The digital multiplexed interface feature supports two signaling techniques: bit-oriented
signaling and message-oriented signaling for direct connection to host computers.
Digital multiplexed interface offers two major advantages:
•
Digital multiplexed interface delivers a standard, single-port interface for linking host
computers internally and externally through a T1 carrier.
•
Since it is compatible with ISDN standards and is licensed to numerous equipment
manufacturers, digital multiplexed interface promotes multi-vendor connectivity.
•
Communication Manager supports two versions of digital multiplexed interface, each
differing in the way information is carried over the 24th channel:
•
Bit-oriented signaling
• Message-oriented signaling
Digital multiplexed interface bit-oriented signalling carries framing and alarm data and
signalling information for connections to host computers and other vendor equipment.
Digital multiplexed interface message-oriented signalling, fully compatible with ISDN-PRI,
uses the same message-oriented signalling format -link access procedure on the D-channel as ISDN-PRI for control and signalling. These signalling capabilities extend the advantages of
digital multiplexed interface message-oriented signalling multiplexed communications to the
public ISDN network.
Direct Inward Dialing (DID) trunks connect Communication Manager to the local central office
for incoming calls dialed directly to stations without attendant assistance.
Traditionally, Central Office (CO) trunks and Direct Inward Dialing (DID) trunks interface an
attendant console with a central office. A CO trunk services outgoing calls and accepts
incoming calls that are terminated at the attendant. A Direct Inward/Outward Dialing (DIOD)
trunk is used for calls that need to be terminated without an attendant interaction.
See E&M signaling -continuous and pulsed.
This screen administers the Centralized Automatic Message Accounting (CAMA) trunks and
provides Caller Emergency Service Identification (CESID) information to the local enhanced
911 system through the local central office.
Foreign Exchange (FX) trunks connect Communication Manager to a Central Office other
than to the local office.
Gives you access to a variety of public and private network services and facilities. The ISDN
standard consists of layers 1, 2, and 3 of the Open System Interconnect (OSI) model.
Systems running Communication Manager can be connected to an ISDN using standard
frame formats: Basic Rate Interface (BRI) and the Primary Rate Interface (PRI).
An ISDN provides end-to-end digital connectivity and uses a high-speed interface that
provides service-independent access to switched services. Through internationally accepted
standard interfaces, an ISDN provides circuit or packet-switched connectivity within a network
and can link to other ISDN supported interfaces to provide national and international digital
connectivity.
The user side will support automatic TEI assignment by the network. Both fixed and automatic
TEI assignment will be supported on the network side.
Enables a single ISDN-PRI trunk group to carry calls to a variety of services, rather than
requiring each trunk group to be dedicated to a specific service. It allows you to set up various
voice and data services and features for a particular call.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Trunk Connectivity Features
Trunk Connectivity
ETSI functionality
The full set of ETSI public-network and private-network ISDN features is officially supported.
This includes Look-Ahead Interflow (LAI), look-ahead routing, and usage allocation.
Also included is all QSIG supplementary services, such as:
•
Name identification
•
Call diversion (including rerouting)
•
Call transfer
• Path Replacement
ETSI functionality does not include:
Facility and non-facility
associated signaling
Feature plus
ISDN-Basic Rate
Interface
•
DCS
•
Non-facility associated signaling
•
D-channel backup
• Wideband signaling
Facility and non-facility associated signaling allows an ISDN-PRI DS1/E1 interface D-channel
to carry signaling information for B-channels (voice or data). D-Channel Backup can also be
administered to increase system reliability.
Feature plus enables those users without DID service to direct dial users on a remote PBX
through the public network. ISDN feature plus eliminates the need for attendant intervention
for those without DID capabilities.
Enables connection of the system to equipment or endpoints that support an Integrated
Services Digital Network (ISDN) by using a standard format called the Basic Rate Interface
(BRI). This feature is a 192-Kbps interface that carries two 64-Kbps B-channels and one 16Kbps D-channel.
ISDN is a global access standard that uses a layered protocol. It eliminates the need for
multiple, separate access arrangements for voice, data, facsimile, and video services and
networks. Using the same pair of wires that carry simple telephone calls, ISDN can deliver
voice, data, and video services in a digital format.
The ISDN-BRI Trunk circuit pack allows Communication Manager to support the T interface
and the S/T interface as defined by ISDN standards (ITU-T recommendation I.411). The
circuit pack provides eight ports to the network and supports two B channels and one D
channel.
The ISDN-BRI Trunk provides the following advantages:
•
Provides an inexpensive way to connect to ISDN services provided by the network
provider
•
Meets almost all ETSI Country protocol requirements
•
Supports essential (not supplementary) ISDN services
•
Multiple subscriber
number - limited
BRI trunks support public-network access outside the U.S. on point-to-midpoint
connections, with the restriction that Communication Manager must not be
configured in a passive bus arrangement with other BRI endpoints. ISDN-BRI trunks
can be used as inter-PBX tie lines using the QSIG peer protocol.
The ISDN standard MSN feature lets customers assign multiple extension to a single BRI
endpoint. The MSN feature works with BRI endpoints that allow the channel ID IE to be
encoded as "preferred."
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Trunk Connectivity Features
Trunk Connectivity
NT interface on TN556C
Presentation restriction
Wideband switching
Multi-Frequency Packet
signaling -Russia
National private
networking support Japan
Personal Central Office
Line
Release Link Trunks
Remote access trunks
Tie trunks
Timed automatic
disconnect for
outgoing trunk calls
Wide Area
Telecommunications
Service
Communication Manager supports the NT (network) side of the T interface using the TN556C
circuit pack. This gives the switch full tie trunk capability using BRI trunks. Communication
Manager supports leased BRI connections through the public network, with a TN2185 on
each end of the leased connection. Communication Manager will not, however, allow
customers to administer both endpoints and trunks on the same TN556C circuit pack.
Restricts the display of calling/connected numbers over ISDN trunks. ISDN trunk groups can
be administered to control the display of calling/connected numbers. Each trunk group can be
administered to display "presentation restricted," "number no available due to networking," or
an administered text string instead of the calling/connected number.
Provides the ability to dedicate two or more ISDN B-channels or DSO endpoints for
applications that require large bandwidth. Certain applications, such as video conferencing
and high-speed data transmission, require extra bandwidth and it becomes necessary to put
several ISDN-PRI narrowband channels into one wideband channel to accommodate the
needs of these applications.
This feature supports both European and North American standards.
See Multi-Frequency Packet signaling.
See National private networking support.
Provides a dedicated trunk circuit between multi-appearance telephones and a CO or other
switch via the network.
Release Link Trunks (RLT) are used between switch locations to provide centralized
attendant service or automatic call distribution group availability.
Tie trunks carry communications between Communication Manager and other switches in a
private network. Several types of trunks can be used, depending on the type of private
network you establish.
This feature provides the capability to automatically disconnect an outgoing trunk call after an
administrable amount of time. The amount of time that can elapse before the trunk is dropped
can be specified, and can vary between 2 and 999 minutes. If the timer field is blank (the
default value), the feature is disabled and the trunk will not be automatically disconnected.
Timed call disconnection applies to all outgoing trunk calls initiated by a party belonging to a
specified Class of Restriction (COR).
Prior to disconnecting the trunk, warning tones are applied to all parties on the call. The first
warning tone occurs when one minute remains on the call. The second warning tone occurs
when 30 seconds remain on the call.
Wide Area Telecommunications Service (WATS) trunks allow you to place long-distance
outgoing voice-grade calls to telephones in defined service areas. The calls are priced
according to distance in the service area, length of the call, time of day, and the day of the
week.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Public Networking and Connectivity Features
Public Networking and Connectivity
Caller ID on analog
trunks
Caller ID on digital
trunks
DS1 trunk service
Echo cancellation with UDS1 circuit pack
E1
T1
Flexible billing
Local exchange trunks
800-service trunks
Central Office trunks
Digital Service 1
trunks
Direct Inward Dialing
trunks
Direct Inward/Outward
Dialing trunks
Foreign Exchange
trunks
Wide Area
Telecommunications
Service
Caller ID on analog trunks allows the system to accept calling name information from a Local
Exchange Carrier (LEC) network that supports the Bellcore calling name specification. The
system can send calling name information in the format if Bellcore calling name ID is
administered.
In the United States, the telephone of a user displays calling party information (if the
telephone is a display telephone). Name and calling number are available from the US central
offices.
This feature may be used in countries that comply with either US. The display of name and
number will work with all Communication Manager digital telephones (DCP and BRI)
equipped with a 40-character or a 32-character alphanumeric display.
See DS1 trunk service.
See Echo cancellation -with UDS1 circuit pack.
See E1.
See T1.
See Flexible billing.
Local exchange trunks connect Communication Manager to a central office. The following
local exchange trunks are some of the types available.
800-service trunks let your business pay the charges for inbound long-distance calls so that
callers can reach you toll-free.
See Central Office.
See DS1 trunk service.
See Direct Inward Dialing.
See Direct Inward/Outward Dialing.
See Foreign Exchange.
See Wide Area Telecommunications Service.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Intelligent Networking Features
Intelligent Networking
Avaya VoIP
Monitoring Manager
Distributed
Communications
System protocol
Attendant with DCS
Direct trunk group
selection
Display
DCS automatic circuit
assurance
DCS over ISDN-PRI Dchannel
DCS protocol -Italy
DCS with reroute
QSIG/DCS voice mail
interworking
Electronic Tandem
Network
Automatic alternate
conditional routing
Trunk signaling and
error recovery
See Avaya VoIP Monitoring Manager.
The Distributed Communications System (DCS) protocol allows you to configure two or more
switches as if they were a single, large system. DCS provides attendant and voice-terminal
features between these switch locations. DCS simplifies dialing procedures and allows
transparent use of some of the Communication Manager features. (Feature transparency
means that features are available to all users on DCS regardless of the switch location.)
See Direct trunk group selection.
See Display.
See DCS automatic circuit assurance.
Enhances DCS by allowing access to the public network for DCS connections between DCS
switch nodes. With this feature (also known as DCS Plus or DCS+), DCS features are no
longer restricted to private facilities. The ISDN-PRI B-channel is used for voice
communications, and the ISDN-PRI D-channel is used to transport DCS control information.
See Distributed Communications Systems protocol.
Allows a DCS call to be rerouted over a different path if the switch finds a better quality and
lower cost route. This feature allows for rerouting the call after a transfer or rerouting during a
call. DCS with reroute is similar to the rerouting capabilities used with QSIG.
See QSIG/DCS voice mail interworking.
In an Electronic Tandem Network (ETN) -also known as Private Network Access (PNA) Communication Manager provides a variety of features on a network-wide basis. It allows
calls to other systems in a private network. These calls do not use the public network. Instead,
they are routed over your dedicated facilities.
You can control the routing of particular calls using conditional routing. For example, you can
limit the number of communications satellite hops (communications satellite links used as
trunks) in any end-to-end private network routing pattern. Limiting the number of satellite hops
may be desirable for controlling transmission quality or call delay in both voice and data calls.
The reliability of electronic tandem network calls is improved by allowing a trunk call to be
retried on another circuit when signaling failures occur.
•
tandem switch: A switch within an ETN that provides the logic to determine the best
route for a network call, possibly modifies the digits outpulsed, and allows or denies
certain calls to certain users.
•
tandem through: The switched connection of an incoming trunk to an outgoing trunk
without human intervention.
•
Extension number
portability
Tandem Tie-Trunk Network (TTTN): A private network that interconnects several
customer switching systems.
When employees move within the network, they can retain their extension numbers. The
ability to keep extension numbers, and even electronic tandem network and direct inward
dialed numbers, when moving to other locations within the company eliminates missed calls
and saves valuable time.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Intelligent Networking Features
Intelligent Networking
Internet Protocol
Alternate gatekeeper
and registration
addresses
The capabilities and applications of Communication Manager are extended using IP.
Communication Manager IP supports audio/voice over a LAN or WAN, and it ensures that
remote workers have access to communication system features from their PCs.
Communication Manager also provides standards based control between media server and
media gateways allowing communications infrastructure to be distributed to the edge of the
network.
The Communication Manager IP engine offers features that enables users to increase the
quality of voice communications. The Quality of Service (QoS) feature enables users to
administer and download the differentiated services type-of-service value to optimize voice
quality. The QoS feature reduces latency by implementing buffers in the audio-processing
board, and assists some routers in prioritizing audio traffic.
Communication Manager IP also includes hairpin and IP-IP direct connections, two features
that make voice communications more efficient. These features increase the efficiency of
voice communications by reducing both per port costs and IP bandwidth usage.
IP solutions supports trunks, IP communications devices, IP port networks, and IP control for
media gateways. IP solutions is implemented using various IP-media processor circuit packs
inside the servers or the Avaya media gateways. The IP media processors provides H.323
trunk connections and H.323 voice processing for IP telephones. The features that use the IP
media processor also require the CLAN circuit pack or native processor ethernet connectivity.
The IP LAN can also connect through VPN and WAN facilities to extend the customer IP
network across geographically disparate locations. Distributed communication services
(DCS+), or QSIG services, can extend feature transparency, centralized voice mail,
centralized attendant service, call center applications, and enhanced call routing across IP
trunks. Note: To maximize voice quality using IP, you must consider both your hardware and
network configurations. For example, with IP softphones, you can send the audio over
traditional circuit switch lines, providing high quality voice, or over IP using LAN connections.
The IP network must be a switched ethernet infrastructure and have the appropriate
engineering to accommodate bandwidth, latency and packet loss requirements to effectively
provide for real-time voice over IP traffic.
When an IP endpoint (including softphones, IP telephones, and Avaya R300) registers with
the switch, the switch sends back an IP registration address. The switch sends a different IP
address for each registration according to a cyclic algorithm.
If registration with the original CLAN circuit pack IP address is successful, the switch sends
back the IP addresses of all the CLAN circuit packs in one network region, not including
interconnected regions. These CLAN addresses are called gatekeeper addresses. These
addresses can also be used if the call signaling on the original CLAN circuit pack fails. Note:
On switches using the LAN region based on IP Address feature, it is likely that the network
region number assigned to an IP telephone would be different from the network region
number of the TN799 that the telephone is registering through. That difference would mean
the list of TN799 addresses in the same network region as the IP telephone would be empty.
The alternate gatekeeper feature would send a blank list to the IP telephone.
To prevent that from happening, an IP terminal registers with Communication Manager.
Communication Manager then sends to the endpoint the IP addresses of the CLANs in the
same region as the terminal, followed by network regions interconnected with the network
region of the terminal.
If the network connection to one CLAN circuit pack fails, the IP endpoint re-registers with a
different CLAN. Alternate gatekeeper and registration addresses, and CLAN circuit pack load
sharing, spread IP endpoint registration across more than one CLAN circuit pack, increasing
performance and reliability.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Intelligent Networking Features
Intelligent Networking
Classless Interdomain
Routing
Multiple network
regions per CLAN
Multiple location
support for network
regions
Network regions
Quality of Service
802.1p/Q
802.1x multi supplicants
Classless Interdomain Routing (CIDR) is a redefinition of the subnet mask, allowing for the
aggregation of contiguous classful networks under a single network definition. This allows for
more efficient routing table management when administering IP address on Communication
Manager.
Multiple network regions per CLAN enables a single CLAN to provide registration and call
control to IP endpoints in multiple network regions. Communication Manager implements this
approach by allowing IP addresses to be mapped to network regions in a mapping screen,
instead of just to a CLAN. When an IP telephone registers, the switch determines the
telephone’s network region number based on the telephone’s IP address.
Multiple location support for network regions allows remote Avaya media gateways connected
to a central Avaya media server to retain:
•
Local user time
•
Local ARS public analysis tables for local trunking
• Automatic daylight savings time
Local touch tone receivers for IP communications devices, such as Avaya IP telephones.
Communication Manager allows administrators to map locations to IP network regions.
Network regions provide the administrative foundation on which Communication Manager
features are allocated to IP endpoints. A network region is a collection of IP endpoints and
switch IP interfaces interconnected by an IP network.
Endpoints that share network regions typically represent users with common interests. For
example, a customer might have two separate small campuses in a large metropolitan area,
interconnected by a WAN, and both served by the same server running Communication
Manager. Communication Manager allows the customer to define a network region for each
campus, and associate each of their CLAN and IP media processor circuit packs with these
regions.
By employing a variety of Quality of Service (QoS) features, Communication Manager
provides the best possible end-to-end audio experience when all or part of the audio path is
carried over packet facilities. "Best" in this context is defined by the customer as represented
by the system administrator, and represents a trade-off between audio reproduction quality,
audio path delay (latency), audio loss, and network resource consumption.
IEEE standard 802.1Q and 802.1p provide the means to specify both a Virtual LAN (VLAN)
and a frame priority at layer 2 for use by LAN hubs, or bridges, that can do routing based on
MAC addresses. 802.1p/Q provides for 8 levels of priority (3 bits) and a large number (12 bits)
of VLAN identifiers. The VLAN identifier at layer 2 permits segregation of traffic to reduce
traffic on individual links. Because 802.1p operates at the MAC layer, its presence may vary
from LAN segment to LAN segment within a single network region. Flexibility requires that
802.1p/Q options be administered individually for each network interface.
Multi-supplicants are common in IP telephony where PC and IP endpoints are attached to the
same port. For better security and to reduce interdependency between the PC and IP
endpoints, the multi-supplicants mode enables each supplicant to independently authenticate
itself to gain access to the network. The multiple supplicants feature is currently supported on
the G350 and G250 platforms.
In remote sites, the multi-supplicants mode provides:
•
An extra level of security by restricting access only to known users and devices
•
Consistency of security features offered in the gateways’ LAN interfaces in case of
multi-vendor networks (Avaya gateways and Extreme switches)
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Intelligent Networking Features
Intelligent Networking
Camp-on/Busy-out
Call Admission
Control bandwidth
management
CLAN load balancing
Codecs
Differentiated services
Dynamic jitter buffers
Integration with Cajun
rules
A camp-on/busy-out command is commonly used by system technicians to busy-out system
resources that need maintenance or repair. Without it, all active calls using those resources
are indiscriminately dropped if the resource is physically removed from the system. This
disruptive action causes problems for customers, especially when a large number of calls are
torn down.
The Camp-on/Busy-out feature for Prowler, MedPRO, and Cruiser adds the ability to remove
idle VoIP resources from the system pool of available VoIP resources. Note: This feature is
not supported by the G700 or G350 Media Gateway platforms.
The Camp-on/Busy-out feature enables the user to select the media processor to be busiedout while the media processor is still in service. After a call ends that was using resources
within the specified media processor, the idled resource is removed from the system pool of
available resources. Once all of the media processor resources are in a "busy-out" state, the
associated board can be removed from the system without disrupting active calls.
In order to ensure Quality of Service for Voice over IP calls, there is a need to limit overall
VoIP traffic on WAN links. The Call Admission Control (CAC) Bandwidth Management feature
of Communication Manager allows the customer to specify a VoIP bandwidth limit between
any pair of IP network regions. The feature then denies calls that need to be carried over the
WAN link that exceed that bandwidth limit.
CLAN load balancing is the process of registering IP endpoints to CLAN circuit packs
(TN799x). Load balancing occurs among CLANs within a network region.
IP endpoint registration among CLAN circuit packs is done through an algorithm. This
algorithm tracks the number of sockets being used per TN799x circuit pack, and registers IP
endpoints to the TN799x with the most available (unused) sockets. This algorithm applies to
H.248, H.323 signaling groups, H.323 stations, and SIP endpoints. Sockets used by adjuncts
are not included in the socket count.
A codec (coder/decoder) provides the means by which audio is compressed. A codec is
typically used in VoIP. Codecs supported by Communication Manager include G.711, G.723,
and G.729.
With the Differentiated Services (DiffServ) option, the system administrator can administer (by
region) and download, to the TN2302AP, the DiffServ Type-Of-Service (TOS) value. This
allows data networking equipment to prioritize the audio stream at the IP level to promote
voice quality. DiffServ makes use of the TOS octet in the existing IP version 4 header. As
such, it may be set by information senders and used by IP (layer 3) routers within the network.
Propagation delay and jitter is caused when a human’s voice is sampled, encoded, and
packetized for transport over the IP network, but is received and decoded at different rates.
Jitter buffers are used to buffer the audio output to smooth the conversions. Communication
Manager provides dynamic jitter buffers to balance both delay of conversation and rapid
bursts that may occur.
Cajun rules provide a central repository for QoS parameters and allows comprehensive QoS
management across routers, switches, and endpoints. QoS parameters and policies are
assigned according to network regions on a network region and are distributed through
enterprise directory gateway to Communication Manager and to routers and switching
devices.
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IP overload control
QoS for call control
QoS for VoIP
QoS to endpoints
Resource Reservation
Protocol
This enhancement more effectively manages processor occupancy overload situations. The
enhancement applies selected overload mechanisms at a lower occupancy threshold in an
effort to avoid more serious symptoms experienced at higher occupancy levels.
The IP overload control enhancement:
•
fortifies the system against bursts of registration traffic
•
provides a mechanism to alert the far-end to abstain from issuing registrations for
some specified period of time
•
records the event to maintain a history of potential performance problems
•
optimizes the maximum number of simultaneous registrations the server can handle
based on the available memory and CPU cycles
• reduces the frequency that a server might go into overload due to network problems
Communication Manager allows QoS for the signaling packets coming from gatekeepers such
as the CLAN by employing the same standards based DiffServ and 802.1p/Q schemes as
with audio channels. This QoS services further improve the users VoIP audio experience.
Communication Manager implements QoS through the selection of audio codec such as
G.711, G.723 and G.729, and by requesting network prioritization through the layer 3
differentiated services (DiffServ) scheme, as well as the layer 2 IEEE 802.1p/Q prioritization.
DiffServ and 802.1p/Q are supported on voice packets coming to/from the gateway, all the
way down to the endpoints such as IP telephones. Dynamic jitter buffers are also used.
Users can set operating parameters to optimize the audio performance, or quality of service
(QoS), on calls made over your IP network. These parameters include the audio codec,
network priority through DiffServ capability, and the IEEE 802.1p/Q MAC-layer prioritization
and segregation.
Default QoS parameters are downloaded to the IP telephone R1.5 and the IP softphone R3
when the values are not provided by the endpoint installer or the user. Certain options can be
set locally by the endpoints or through the gatekeeper. The endpoints receive the parameters
when the endpoints register, and once they are registered, whenever the administered values
of the QoS parameters are modified.
Resource Reservation Protocol (RSVP) is a QoS signaling protocol. RSVP provides a means
of specifying the requirements of IP packet flow, and determining if the intervening network
can provide the resources to protect that flow through a process called "admission control."
RSVP protection of VoIP audio streams on WANs and other links that are susceptible to
congestion can safeguard the quality of VoIP calls already in progress.
•
IP telephones or gateways request the network routers to reserve bandwidth.
•
The routers act upon the request to allocate bandwidth according to the QoS
request.
•
Sending and receiving
faxes over IP
When the bandwidth is reserved, the call is protected against other network traffic in
a loaded or congested network, thereby ensuring good voice quality.
Administrators can now configure RSVP settings in Communication Manager. When the
RSVP enable field in the IP Network Region screen is set to y, the RSVP Reservation
Parameters field appears.
Starting with Communication Manager release 2.1, users can send and receive faxes over the
voice over IP (VoIP) and modem over IP (MoIP) networks. The firmware that is resident on
the TN2302AP circuit packs (Hardware Vintage 10 or later), the MM760 Media Module, the
G700 Media Gateway, and the G350 Media Gateway, actually performs the processing
necessary to allow proper handling of faxes over an IP network.
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Modem over IP
Relay mode
Pass through mode
T.38 faxes over the
Internet
The modem over IP (MoIP) feature allows for transport of data over a 64kbps unrestricted
clear channel. Starting with Communication Manager release 3.0, when a clear channel data
call is originated, the system communicates to the media processor or VoIP engine to allow a
64kpbs clear channel to be opened for transport.
In relay mode, the firmware detects fax tones and uses the appropriate modulation protocols
(V.xx) to terminate or originate the fax so that the fax can be carried over the IP network. To
reduce bandwidth over the IP network, the system encodes the modulated analog signal from
the fax, and uses a relay coder/decoder. This process improves the reliability of transmission.
Also, because the data packets for faxes in relay mode are sent almost exclusively in one
direction, from the sending endpoint to the receiving endpoint, bandwidth use is reduced.
Relay mode works only if the receiving fax endpoint and the sending fax endpoint both
communicate through Avaya Communication Manager media servers. This transport of fax
signals occurs at a 9600 bps rate (though this rate may vary with the version of firmware).
This mode may be used for fax calls to and from Communication Manager R2.0 systems.
V.32 modem relay (With CM 3.1)
V.32 modem relay is an option that provides a low-bandwidth solution for secure voice
terminals on the TN2602AP circuit pack. For customers wishing to use standard data
modems, modem pass thru is the appropriate solution. Both modem pass thru and V.32
modem relay already exist on the TN2302AP circuit pack, so it is now possible for these two
circuit packs to interoperate.
Alternatively, you can choose to have fax signals sent in "pass through" mode. Pass through
mode means the fax signals are transported using G.711-like encoding and are delivered to
the receiving fax endpoint as IP signals. This capability provides higher quality transmission in
the circumstance where endpoints in the network are all synchronized to the same clock
source. Pass through mode works only if the receiving fax endpoint and the sending fax
endpoint both communicate through Avaya Communication Manager media servers.
The transport speed is up to the equivalent of circuit-switched calls and supports G3 and
Super G3 fax rates, up to and including 33.6 kbps.
With Communication Manager, Release 2.1, users can send and receive faxes over the VoIP
network using the T.38 standard for relay. The firmware resident on the TN2302AP circuit
packs (Hardware Vintage 10 or later), the MM760 Media Module, the G700 Media Gateway,
and the G350 Media Gateway actually performs the processing necessary to allow proper
handling of faxes over an IP network. This transport of fax signals occurs at a 9600 bps rate.
The T.38 fax capability allows users to send faxes to and receive faxes from endpoints that
are connected to non-Avaya systems. This capability is standards-based and uses IP trunks
and H.323 signaling to allow communication with non-Avaya systems. Additionally, the T.38
fax capability uses the UDP protocol. Note: Fax endpoints served by two different Avaya
media servers can also send T.38 faxes to each other if both systems are enabled for T.38
fax. In this case, the media servers also use IP trunks.
However, if the T.38 fax sending and receiving endpoints are on port networks or media
gateways that are registered to the same media server, the gateways or port networks revert
to Avaya fax relay mode. Avaya fax relay mode is more efficient that T.38 from a bandwidth
perspective.
Both the sending and receiving systems must announce support of T.38 fax data applications
during the H.245 capabilities exchange. Avaya systems announce support of T.38 fax if the
capability is administered on the Codec Set screen for the region and a T.38-capable media
processor was chosen for the voice channel. In addition, for a successful fax transmission,
both systems should support the H.245 null capability exchange (shuffling) in order to avoid
multiple IP hops in the connection. Note: The T.38 fax capability does not support TCP.
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Pass through mode
Support of T.38 fax
relay
Encryption
Shuffling and
hairpinning
Variable length ping
Variable Length
Subnet Mask
NAT with shuffling
You can assign packet redundancy to T.38 standard faxes to improve packet delivery and
robustness of fax transport over the network.
You cannot send faxes in pass through mode with the T.38 standard.
T.38 fax relay over IP is now supported with the TN2602AP IP Media Resource 320 circuit
pack.
The T.38 fax call set up may initiate as a G.711, but once fax tones are detected, traffic is
encoded/decoded using the T.38 specification.
In the case of duplicated TN2602 circuit packs that are in an active-standby mode, the system
sends this information to the standby circuit pack when an interchange occurs.
You can encrypt fax pass through calls using either Avaya Encryption Algorithm (AEA) or
Advanced Encryption Standard (AES). You can encrypt fax relay calls with AEA only.
Shuffling and hairpinning can improve traffic handling performance and improve voice quality
by more efficiently using both Communication Manager switching fabric by allocating, when
possible, available IP network resources.
"Shuffling" means rerouting the audio channel connecting two IP endpoints. After shuffling,
the audio which previously was carried in a mixed connection of IP signaling and TDM bus
signaling, goes directly through the LAN or WAN between the two IP endpoints. Shuffling also
can mean reversing this process if an endpoint requests a resource to support a feature, such
as conferencing that requires the TDM bus.
"Hairpinning" means rerouting the audio channel connecting two IP endpoints so that the
bearer (audio) packets are routed through the TN2302AP IP Media Processor board in IP
format, without having to go through the IP to TDM conversion or traverse the TDM bus.
Provides an enhancement to the ping command included in R7.1. This enhancement
specifies a longer packet to be sent by ping and shows if a router or host has a problem
fragmenting or integrating transferred packets.
Variable Length Subnet Mask (VSLM) is a redefinition of the subnet mask, allowing for a more
efficient allocation of IP addresses within a traditional classful block when administering IP
address on Communication Manager.
Communication Manager allows IP endpoints to shuffle if they are behind a Network Address
Translation (NAT) device in an IP network. Note: Network Address Translation (NAT) is a
method to address the shortage of IP V4 addresses by allowing globally register IP addresses
to be reused by native networks. A NAT device translates between translated and native IP
addresses.
Communication Manager supports IP direct calls (a call that has been shuffled) between two
IP endpoints that are translated through a NAT device.
This enhancement works with static one-to-one NAT. It does not facilitate Port Address
Translation (PAT), also known as Network Address Port Translation (NAPT). This
enhancement does not work with many-to-one NAT.
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TTY
People with hearing or speech disabilities often rely on a device known as a TTY in order to
communicate on telephone systems. The term "TTY" is an abbreviation for Teletypewriter.
The term "TDD" (Telecommunication Device for the Deaf) is also frequently used. The term
TTY is generally preferred, however, because many people who use these devices are not
deaf.
TTY devices typically resemble small laptop computers, except that there is a one- or two-line
alphanumeric display in place of the computer screen.
Connection to the telephone network is generally through an acoustic coupler into which the
user places the telephone’s handset, or through an analog RJ-11 tip/ring connections.
Reliable transmission of TTY signals is supported by Communication Manager. This complies
with the requirements and guidelines outlined in United States accessibility-related laws.
Those laws include:
•
Titles II, III, and IV of the Americans with Disabilities Act (ADA) of 1990.
• Sections 251 and 255 of the Telecommunications Act of 1996.
Section 508 of the Workforce Investment Act of 1998. Communication Manager TTY support
is currently restricted to TTY devices that use the:
•
US English standard TTY protocol, specified by ANSI/TIA/EIA 825 as: "A 45.45 Baud
FSK modem."
• UK English standard TTY protocol, Baudot 50.
Important characteristics of this standard are:
TTYs are silent when not transmitting. Unlike fax machines and computer modems, TTYs
have no "handshake" procedure at the start of a call, nor do they have a carrier tone during
the call. This approach has the advantage of permitting TTY tones, DTMF, and voice to be
intermixed on the same call. Note: A large percentage of people who use TTY devices
intermix voice and typed TTY data on the same call. The most common usage is by people
who are hard of hearing, but nevertheless able to speak clearly. These people often prefer to
receive text on their TTY device and then speak in response. This process is referred to as
Voice Carry Over (VCO).
•
Operation is "half duplex." TTY users must take turns transmitting and typically
cannot interrupt each other. If two people try to type at the same time, their TTY
devices might show no text at all or show text that is unrecognizable. Also, there is
no automatic mechanism that lets TTY users know when a character they have
correctly typed has been received incorrectly.
•
Each TTY character consists of a sequence of seven individual tones. The first tone
is always a "start tone" at 1800 Hz. This is followed by a series of five tones, at either
1400 or 1800 Hz, which specify the character. The final tone in the sequence is
always a "stop tone" at 1400 Hz. The stop tone is a border that separates this
character from the next.
The following types of systems support TTY communication:
•
Analog telephones and trunks
•
Digital telephones and trunks
•
VoIP gateways
•
Messaging systems
•
Automated attendant systems
• IVR systems
Wireless systems in which a TTY-compatible coder is used As long as the user’s TTY device
supports the following, Communication Manager allows:
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TTY over analog
and digital trunks
TTY over Avaya IP
trunks
TTY relay mode
TTY pass through
mode
•
Voice and TTY tones to be intermixed on the same call.
•
DTMF and TTY (with or without voice) to be intermixed on the same call. This allows
TTY users to access DTMF-based voice mail, auto-attendant, and IVR systems.
• The use of acoustically coupled and "direct connect" (RJ-11) TTY devices.
Communication Manager supports TTY calls within a gateway or port network between two
analog telephones. TTY calls from a gateway or port network over analog trunks or digital
trunks is also supported.
Communication Manager supports calls over IP trunks, as well as Inter-Gateway Calls (IGC).
Note: For this feature to work, both the sender (near end) and the receiver (far end) of a TTY
call must each be connected to Avaya IP trunks. This feature does not work if either
telephone is an IP telephone.
In relay mode, the system:
•
detects TTY characters
•
transports a representation of the characters over the IP network
• regenerates TTY characters/tones for delivery to the TTY device
This transport of TTY supports US English TTY (Baudot 45.45) and UK English TTY (Baudot
50). TTY uses RFC 2833 or RFC 2198 style packets to transport TTY characters.
Depending on the presence of TTY characters on a call, the transmission toggles between
voice mode and TTY mode. The system uses up to 16 kbps of bandwidth when sending TTY
characters, and normal bandwidth of the audio codec for voice mode. This mode may be used
for TTY calls to and from Communication Manager R2.0 systems.
In relay mode, you can also assign packet redundancy. Packet redundancy means the media
gateways send duplicated TTY packets to ensure and improve quality over the network.
Alternatively, you can choose to have TTY signals sent in pass through mode. With pass
through mode enabled, when the system detects TTY characters, the system uses G.711
encoding to transport the TTY signals end-to-end over the IP network. G.711 encoding pass
through mode means the TTY signals are changed to digital format, and are delivered to the
receiving endpoint after unencoding the digital signals.
Pass through mode provides higher quality transmission when endpoints in the network are
all synchronized to the same clock source.
In pass through mode, you can also assign packet redundancy. Packet redundancy means
the media gateways send duplicated TTY packets to ensure and improve quality over the
network.
Pass through mode uses more network bandwidth than relay mode. Pass through TTY uses
87-110 kbps, depending on the packet size, whereas TTY relay uses, at most, the bandwidth
of the configured audio codec. Redundancy increases bandwidth usage even more.
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Basic
Call completion
Call forwarding
(diversion)
QSIG
QSIG provides compliance to the International Standardization Organization (ISO) ISDN-PRI
private-networking specifications. QSIG is defined by ISO as the worldwide standard for
private networks. QSIG features are supported on BRI trunks.
QSIG is the generic name for a family of signaling protocols. The Q-reference point or
interface is the logical point where signaling is passed between 2 peer entities in a private
network. QSIG signaling can provide feature transparency in a single-vendor or multi-vendor
environment.
QSIG provides call-related supplementary services. These are services that go beyond voice
or data connectivity and number transport and display. Examples of supplementary services
include name identification, call forwarding (diversion), and call transfer.
Call completion utilizes the QSIG platform enhancement call independent signaling
connections and is functionally equivalent to the Distributed Communications System (DCS)
feature: auto-callback. The call completion feature includes a connection release method. The
connection release method clears the Temporary Signaling Connection (TSC) after each
phase of call-independent signaling and establishes a new TSC for each subsequent phase.
QSIG call forwarding (diversion) is based on the Communication Manager call forwarding
feature. It extends the feature transparency aspects of call forwarding over a QSIG trunk:
•
If QSIG call forwarding is activated, all calls are diverted immediately.
•
If QSIG call forwarding with busy/do not answer is activated and a station is busy, a
call is diverted immediately.
•
Call Independent
Signaling Connections
Call offer
If QSIG call forwarding with busy/do not answer is activated and a station is idle but
the call is not answered, a call is diverted after a specified number of rings.
These features are activated either by dialing a Feature Access Code (FAC) or by pressing a
button.
Call Independent Signaling Connections (CISC) are used to pass QSIG supplementary
service information that is independent of an active call between two QSIG compliant nodes.
Implementation is based on the ISO standard for CISC. It is possible to determine the status
of a QSIG TSC by using the "status signaling group" command on the SAT.
This feature, on request from the calling-user (or on behalf of that user), enables a call to:
•
Be offered to a busy called-user
•
Call transfer
Called name ID
Centralized Attendant
Service
Attendant display of
Class of Restriction
Wait for a busy called-user to accept the call when the necessary resources have
become available
QSIG call transfer differs from the standard Communication Manager transfer feature in that
additional call information is available for the connected parties after the transfer completes.
However, the information is only sent for QSIG trunks. If one call is local to the transferring
switch, that user receives the name of the party at the far end.
The QSIG called name feature presents the name of the called party on the display of the
calling party while the call is ringing. It then reverts to "connected name" when answered.
Provides you with the capability to have all your attendants in one location, serving users in
multiple locations. QSIG CAS does not utilize separate Release Link Trunks (RLT). This
feature will not restrict calls from going out over non-QSIG trunks; however, the full
functionality of the QSIG CAS will not be available.
While on a call, the attendant can press a "COR display" button to see the class of restriction
of the user. The attendant will not block the transfer of the restricted line to the user. This
feature is used for informational purposes only.
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Attendant return call
Priority queue
QSIG path
optimization simplified
QSIG redirection
display is
administrable
Rerouting and path
replacement by trunk
group
If a call that is transferred by the attendant goes unanswered for a specified period of time,
the call is returned to the attendant. Preferably the call will transfer back to the attendant who
transferred the call.
QSIG MSI will pass more information to the main PBX. This information enables calls coming
in from a QSIG CAS branch to be placed in the appropriate place in the queue, as if the call
originated on the main PBX.
Improvements were made to the dial plan to simplify path replacement and diversion with
rerouting.
When a Do-Not-Call (DNC) server authorizes a call through QSIG, the server returns the
routing number to the CM using QSIG Redirection. The word “forward” and the secret routing
number were displayed on the caller’s telephone display.
QSIG redirection is now administrable so a customer can turn off the information from the
caller’s telephone display if they choose.
You can now administer individual QSIG trunk groups not to use rerouting and path
replacement, while leaving these capabilities active for other trunk groups. The Trunk Group
screen now provides a new page containing two new fields for this purpose:
•
Diversion By Reroute
• Path Replacement
These two new fields are visible only if the Basic Supplementary Services and the
Supplementary Services With Reroute fields on the Optional Features screen are set to y, and
the Supplementary Service Protocol field on the Trunk Group screen is set to b. The default
value for both fields is y.
•
If you set the Diversion By Reroute field to n, the Call Diversion feature uses forward
switching rather than rerouting.
•
RLT emulation
through a PRI
Communication
Manager/Octel QSIG
integration
Manufacturer-Specific
Information
If you set the Path Replacement field to n, the Path Replacement With Retention and
the Path Replacement Method fields are no longer visible, and the trunk group does
not use path replacement.
ISDN QSIG trunks will route calls from the branch PBX to the main PBX. You no longer have
to specify a dedicated RLT network. The QSIG path replacement takes care of the trunk
optimization. You have the flexibility to route calls to the main PBX.
Communication Manager enables integration of Octel messaging servers through QSIG. See
Octel integration.
QSIG handles non-standardized information that is specific to a particular PBX or network.
This information is known as Manufacturer Specific Information (MSI). A manufacturer can
define manufacturer-specific supplementary services operations after it has:
•
Applied to a sponsoring and issuing organization (ECMA or European Computer
Manufacturers Association in this case)
•
Been assigned an organization identifier. This organization identifier is used as the
root of the manufacturer-specific service-operation value.
All MSI operation values should be unique to that manufacturer.
Manufacturer-specific supplementary services can be created using specific operations
encoded with the identifier of the manufacturer. Communication Manager supports non-QSIG
applications that transport information across QSIG networks in MSI. Applications have the
same functionality over QSIG networks that they have over non-QSIG networks. Applications
that use MSI include Centralized Attendant Service, Transfer to Audix, Best Service Routing,
and QSIG VALU.
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Intelligent Networking
Message Waiting
Indication
Leave Word Calling
Name and number
identification
Path replacement with
path retention
QSIG/DCS voice mail
interworking
Reroute after
diversion to voice mail
Stand-alone path
replacement
Supplementary
services and rerouting
The system indicates that a guest telephone has received one or more messages in their
voice mailbox. An automatic message waiting lamp light at the telephone of the called party.
See Leave Word Calling.
Allows a switch to send and receive the calling number, calling name, connected number,
and connected name. Additional parameters that control the display of the connected name
and number are administered on the Feature-Related System-Parameters screen. QSIG
Name and Number Identification displays up to 15 characters for the calling and connected
name and up to 15 digits for the calling and connected number across ISDN-PRI interfaces.
With this feature, a call between switches in a private network can be replaced with new
connections while the call is active. This feature is invoked when a call is transferred and
improvements may be made in costs.
For example, after a call is transferred, the two parties on the transferred call can be
connected directly and the unnecessary trunks are dropped off the call. The routing
administered at the endpoints may allow for a more cost-effective connection.
Earlier versions of DEFINITY could not route a call over the original trunk when path
replacement was turned on. Path Replacement features Path Retention, which allows
Communication Manager to use the original trunk group path when the routing analysis
performed by the switch shows the original trunk group to be the best route.
QSIG/DCS Voice Mail Interworking is an enhancement to the current QSIG feature. It
integrates DCS and QSIG Centralized Voicemail through the DCS+/QSIG gateway. Switches
labeled DCS+/QSIG integrate multi-vendor PBXs into a single voice messaging system.
QSIG/DCS Voice Mail Interworking works on G3r, G3si, and G3csi. It provides network
flexibility, DCS functionality without a dedicated T1.
Supports path optimization for calls that are diverted to a QSIG voice mail hunt group. That is,
the switch moves the call to the shortest route between the caller and the voice mail system.
For example, if user A on switch A calls user B on switch B and the call goes to a voice mail
system attached to switch C, then the call is using up two trunks: A-B and B-C. If there is a
trunk that directly connects switches A and C, this feature will drop the A-B and B-C
connection and set up a new call from switch A to switch C, thus saving one trunk. The
reroute happens automatically; the user never knows that the extra trunk was dropped.
Path Replacement is the process of routing an established call over a more efficient path,
after which the old call is torn down leaving those resources free. Path Replacement offers
potential savings by routing calls more efficiently, saving resources and trunk usage.
Path replacement can exist as a stand-alone feature, or occur in the following additional
cases:
•
Call Forwarding by Forward Switching supplementary service, including the case
where Call Diversion by Rerouting fails, and Call Forwarding is accomplished via
forward switching
•
Gateway scenarios where Communication Manager, serving as an incoming or
outgoing gateway, invokes PR to optimize the path between the gateways
•
Calls in queue/vector processing even though no true user is on the call yet
• QSIG Lookahead Interflow call, Best Service Route call, or adjunct route
The QSIG standard defines Supplementary Services as those service beyond voice or data
connectivity and number transport and display. Examples include call forwarding, transfer and
call hold.
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Intelligent Networking
Call coverage
Call coverage and
CAS
Increased Classes of
Restriction
Increased text fields
for feature buttons
Increased quantity of
NCA TSCs and FTSCs
Reset IP stations by
subnet enhancement
Distinctive alerting
Uniform Dial Plan
Dial Plan Expansion
VALU
This feature provides similar call coverage as DCS call coverage and Call Coverage Remote
Off Net (C-CRON). The call will come back if covered over QSIG. The functionality will only be
complete when all the switches are running under Communication Manager and using QSIG
VALU. The covered-to party can still receive distinct alerting.
When a trunk has both CAS and VALU Call Coverage activated, the coverage display
information is provided on calls that cover from a branch PBX to the main PBX. Path
replacement will be attempted after coverage.
The Classes of Restriction (COR) feature is increased from a total of 96 possible CORs to
996 possible CORs. Classes of Restriction are numbered from 0 to 999, with four CORs 996, 997, 998, and 999 - reserved by the system. The CORs that are available for the user to
assign are from 0 to 995.
If you are using certain newer phones with expanded text label display capabilities, the
Increase Text Fields for Feature Buttons feature allows you to program and store up to 13character labels for associated feature buttons and call appearances.
This feature is currently available for the 2410 (release 2 or newer) and 2420 (release 4 or
newer) DCP telephones. Support for the newer 46xx IP telephones may be added in the
future.
In an S87x0 Media Server or an S8500 Media Server configuration, you can now have up to
999 NCA TSCs system-wide, and up to 999 NCA TSCs per signalling group. You can also
have up to 250 FTSCs.
Customers can reset telephones in a multi-floor building by subnet, and perform controlled
station resets using the reset ip-stations command to reset by subnet.
Provides distinctive ringing, internal and external, to the remote called party when the call is
routed over the QSIG network.
A unique four- or five-digit number assigned to each station on the network. Uniform
numbering gives each station a unique number (location code plus extension) that can be
used at any location in the electronic tandem network to access that station, Communication
Manager enhances the standard UDP with the unrestricted 5-digit Uniform Dial Plan, which
allows up to five digits to be parsed for call routing.
Communication Manager allows you to expand your dial plan to 6 or 7 digits (from 4-digit or 5digit dial plans). This affects all extensions, including stations, data modules, agent login IDs,
vectors, and so on.
This change increases the total number of extensions that can exist in any dial plan. It also
allows Avaya servers to participate in networks that already use 6-digit or 7-digit dial plans -for
example, a network of switches made by other vendors.
Administrators have the flexibility to administer dial plans between 3 and 7 digits in length,
and Communication Manager supports mixed digit lengths in the same dial plan.
Customers upgrading to Communication Manager can choose to migrate to the 6-digit or 7digit dial plan or not. Customers who choose not to migrate may convert their dial plans at a
later date.
Distributed Communications System (DCS) protocol is limited to a dial plan of 3-5 digits. If
your dial plan requires 6 or 7 digits, QSIG, which is the generic name for a family of signaling
protocols, is required.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Intelligent Networking Features
Intelligent Networking
Multi-location dial
plans
Punctuation on station
displays
When a customer migrates from a multiple voice server QSIG/DCS network to a single voice
server whose gateways are distributed across a data network, it may initially seem as if some
dial plan functions are no longer available.
This feature preserves dial plan uniqueness for extensions and attendants that were provided
in a multiple QSIG/DCS network, but were lost when customers migrated to a single
distributed network. This feature provides dial plan capabilities similar to those provided
before the migration, including:
•
extension uniqueness
•
announcement per location
•
local attendant access
• local ARS code administration
A major reason to migrate customers from a multiple QSIG/DCS environment to a single
S8700 network is to provide a greater set of features and help reduce costs. Migrating to a
single network reduces the number of systems a customer has to maintain. That in turn
lowers administration costs -one switch to administer instead of multiple switches, one dial
plan to maintain, and so on. With a single distributed network solution, some features no
longer work transparently across multiple locations.
For example, in a department store with many locations, each location might have had its own
switch with a QSIG/DCS network. That way, the same extension could be used to represent a
unique department in all stores. For example, extension 123 might be the luggage department
in all stores. If the customer migrates to a single distributed network, this functionality is not
available without this feature.
In addition, an S8700 solution does not assure that a call that is routed to an attendant would
terminate at the local attendant. Let us use an example of a public school district that
previously was networked with a switch at each school. If the school district migrates to an
S8700 network, dialing the attendant access code at your school may not route your call to
the local attendant.
Instead of having to dial a complete extension, the multi-location dial plan feature allows a
user to dial a shorted version of the extension. For example, a customer can continue to dial
4567 instead of having to dial 123-4567. Communication Manager takes the location prefix
and adds those digits to the front of the dialed number. The switch then analyzes the entire
dialed string and routes the call based on the administration on the Dial Plan Parameters
screen.
On digital telephone displays, Communication Manager can display punctuation to make
reading a 6-digit or 7-digit extension easier. The number of digits plus punctuation that can be
displayed cannot exceed eight characters.
Punctuation marks that are allowed include:
•
hyphen (for example, xxx-xxxx)
•
period (for example, xxx.xxxx)
•
space (for example, xx xx xx) Formats for displaying numbers with punctuation are
on the Dial Plan Parameters screen.
•
the default 6-digit extension display format is xx.xx.xx
•
the default 7-digit extension display format is xxx-xxxx
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Intelligent Networking Features
Intelligent Networking
Extended trunk
access
Tildes to hide names
in directory
Used with Uniform Dial Plan, allows the system to send any unrecognized number (such as
an extension not administered locally) to another system for analysis and routing. Such
unrecognized numbers can be Facility Access Codes, Trunk Access Codes, or extensions
that are not in the Uniform Dial Plan table. Non-Uniform Dial Plan numbers are administered
on either the First Digit Table (on the Dial Plan Record screen) or the Second Digit Table.
They are not administered on the Extended Trunk Access Call Screening Table. Extended
Trunk Access helps you make full use of automatic routing and Uniform Dial Plan.
Extension Number Portability -When employees move within the network, they can retain their
extension numbers. The ability to keep extension numbers, and even Electronic Tandem
Network and Direct Inward Dialed numbers, when moving to other locations within the
company eliminates missed calls and saves valuable time.
Display names that begin with a single tilde (~) convert to extended ASCII characters and are
available to the Integrated Directory.
Display names that begin with two tildes (~~) are hidden from the Integrated Directory, but are
not converted to extended ASCII.
Display names that begin with three tildes (~~~) both are hidden from the Integrated Directory
and convert to extended ASCII.
Additional tildes in the display name turn conversion to extended ASCII off again (4, 6, etc.
tildes) and back on (5, 7, etc. tildes).
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Data Interface Features
Data Interfaces
Administered
connections
Data call setup
Data hot line
Data modules
Data privacy
Data restriction
Default dialing
Automatically establishes an end-to-end connection between two access or data endpoints
based on administered attributes. This feature provides capabilities such as alarm notification,
including an administrable alarm type and threshold; automatic restoration of connections
established over a Software-Defined Data Network; ISDN-PRI trunk group [service may be
referred to as ISDN-PRI (AC/AE) Service]; scheduled as well as continuous connections; and
administrable-retry interval for failed connection attempts.
Enables the setting up of data calls using a variety of methods, such as: keyboard dialing,
telephone dialing, Hayes command dialing, permanent switched connections, administered
connections, automatic calling unit interface, and Hot Line dialing. Data Call Setup is provided
for both DCP and ISDN-BRI telephones.
Provides for automatic placement of a data call when the originator hangs up. Data Hot Line
may be used for security purposes. This feature offers fast and accurate call placement to
commonly called data endpoints. Data terminal users who constantly call the same number
can use Data Hot Line to automatically place the call when they hang up the telephone.
Data modules connect systems running Communication Manager with other communications
equipment, changing protocol, connections, and timing as necessary.
Communication Manager supports the following types of data module:
•
High Speed Links
•
Data stands
•
Modular-processor data module
•
7000-series data modules
•
Modular-trunk data module
•
Asynchronous Data Unit
•
Asynchronous Data Module (for ISDN-Basic Rate Interface telephones)
• Terminal adapters
All of these data modules support industry standards and include options for setting the
operating profile to match that of the data equipment.
Data Privacy protects analog data calls from being disturbed by any overriding or ringing
features of the system. Data Privacy is activated when you dial an activation code at the
beginning of the call.
Protects analog data calls from being disturbed by any overriding or ringing features of the
system. It is administered at the system level to selected analog and multi-appearance
telephones and trunk groups.
Provides data terminal users who dial a specific number the majority of the time a very simple
method of dialing that number. This feature enhances Data Terminal (Keyboard) Dialing by
allowing a data terminal user to place a data call to a pre-administered destination in several
different ways, depending on the type of data module. Data Terminal Dialing and
Alphanumeric Dialing are unaffected.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Data Interface Features
Data Interfaces
IP asynchronous links
IP asynchronous links enable Communication Manager to transfer existing asynchronous
adjunct connectivity to an Ethernet (TCP/IP) environment. IP asynchronous links support
switch server applications, as well as client applications. Systems running Communication
Manager can connect to System Management applications such as the Avaya Visibility Suite
over the LAN. Call Detail Recording (CDR) devices, Property Management System (PMS)
and printers can be connected using asynchronous TCP/IP links.
IP asynchronous links:
•
Reduce the cost of connecting to systems running Communication Manager for
various adjuncts
•
Allow for an open architecture to transport information and increases the speed at
which data is transferred
•
Allow customers to manage applications from on-site or remote locations
•
Allow several system management applications to run on a single PC, thereby
reducing hardware requirements
•
Guarantee data delivery through a reliable session-layer protocol
•
Modem pooling
Multimedia application
server interface
Support the existing serial hardware investment of a customer through use of
Network Terminal Servers
Enables switched connections between digital data endpoints (data modules) and analog data
endpoints and acoustic coupled modems. Data transmission between a digital data endpoint
and an analog endpoint requires a conversion since the DCP format used by the data module
is not compatible with the modulated signals of an analog modem. A modem translates DCP
format into modulated signals and vice versa. The Modem Pooling feature provides a set of
modems for such conversions.
Communication Manager modem pools are assigned into modem pool groups. A group can
have up to 32 modems, called "members." Communication Manager can have as many as 63
modem pool groups.
The Multimedia Application Server Interface provides a link between Communication Manager
and one or more Multimedia Communications eXchange nodes. A Multimedia
Communications eXchange is a stand-alone multimedia call processor produced by Avaya.
This link to Communication Manager enhances the capabilities of each Multimedia
Communications eXchange system by enabling it to share some of the Communication
Manager features. In particular, the interface provides the following advantages:
•
Call Detail Recording (CDR). The capture of call detail records so you can analyze
the call patterns and usage of multimedia calls just as Communication Manager
administrators analyze normal calls.
•
Automatic Alternate Routing/Automatic Route Selection (AAR/ARS). The intelligent
selection of the most cost-effective routing for calls, based on available resources
and your carrier preference. The system may select public trunks via DEFINITY
Multimedia eXchange (MMCX).
•
Multimedia calling
Voice Mail Integration. You can access your EMBEDDED AUDIX or INTUITY AUDIX
voice messaging system from a Multimedia Communication eXchange (MMCX).
Multimedia calls are initiated with voice and video only. Once a call is established, one of the
parties may initiate an associated data conference to include all of the parties on the call who
are capable of supporting data. The data conference is controlled by an adjunct device called
an Expansion Services Module (ESM).
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Avaya Communication Manager Feature Overview
Data Interface Features
Data Interfaces
Multimedia call early
answer on vectors and
stations
Early Answer is a feature applied to multimedia calls in conjunction with conversion to voice.
Early Answer:
•
Answers the data call
•
Establishes the multimedia protocol prior to completion of a converted call
•
Multimedia Call
Handling
Multimedia call
redirection to
multimedia endpoint
Multimedia data
conferencing (T.120)
through ESM
Multimedia hold,
conference, transfer,
and drop
Multimedia multipleport networks
Pass advice of charge
information to world
class BRI endpoints
Ensures that a voice path to/from the originator is available when the (voice) call is
answered
For an incoming call, Early Answer answers the dynamic service-link calls when the
destination endpoint answers, unless Early Answer is specified during routing or termination
processing. Note: The "destination voice endpoint" might be an outgoing voice trunk if the
destination voice station is forwarded or covered off-premises.
Multimedia Call Handling (MMCH) enables you to control voice, video, and data transmissions
using your telephone set. The feature buttons on a multi-function telephone enable you to
conduct video conferences, and forward, cover, hold, or park multimedia calls much as you
would a standard voice call. You can also share PC applications so that you and colleagues
can collaborate while working from remote sites.
A dual port multimedia station may be a destination of call redirection features such as call
coverage, forwarding, and station hunting. The station can receive and accept full multimedia
calls or data calls converted to multimedia.
The data conference is controlled by an adjunct device called an Expansion Services Module
(ESM). The Expansion Services Module is used to terminate T.120 protocols [including
Generalized Conference Call (GCC), a protocol standard for data conference control] and
provide data conference control and data distribution. The MultiMedia Interface circuit pack,
TN787, is used to rate adapt T.120 data to/from the ESM.
Station users have the ability to activate hold, conference, transfer, or drop on multimedia
calls. Multimedia endpoints and voice-only stations may participate in the same conference.
Communication Manager supports the equivalent of 580 Basic mode complexes operating at
6CCS traffic level. All enhanced mode complexes operate with soft-mode service links since
the use of hard-mode service links reduces capacities. G3si limits are 1/3 to 1/2 of the G3r
limits, depending on memory limitations and port network limitations.
Provides Advice of Charge (AOC) information to World Class BRI (WCBRI) endpoints. On a
call using a WCBRI endpoint, AOC information will be displayed on the endpoint after the call
has completed and the far end has hung up.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Call Routing Features
Call Routing
Alternate facility
restriction levels
Automatic routing
features
Automatic Alternate
Routing
Automatic Route
Selection
ARS/AAR dialing
without FAC
AAR/ARS overlap
sending
AAR/ARS partitioning
Allows Communication Manager to adjust facility restriction levels or authorization codes for
lines or trunks. Each line or trunk is normally assigned a facility restriction level. With this
feature, Alternate Facility Restriction Levels are also assigned. Attendants can change to the
alternates, thus changing access to lines and trunks. You might want to use this feature to
disable most long-distance calling at night, for example, to prevent unauthorized staff from
making long-distance calls.
Communication Manager provides a variety of automatic routing features for public and
private networks. Automatic Alternate Routing (AAR) and Automatic Route Selection (ARS)
are the foundation for these automatic-routing features. They route calls based on the
preferred (normally the least expensive) route available at the time the call is placed.
Generally, AAR routes calls over a private network and ARS routes calls using the public
network numbering plan. However, both AAR and ARS support public and private networks.
You can use the other features listed in this section when you use AAR and ARS.
Automatic Alternate Routing (AAR) allows private network calls to originate and terminate at
one or many locations without accessing the public network. When you dial an access code
and telephone number, AAR selects the most desirable route for the call and performs digit
conversion as necessary. If the first choice route is unavailable, another route is chosen
automatically.
The numbers you call using AAR are normally private-network numbers. However, you can
call a public-network number, a service code, an international number, operator access
code, or an operator-assisted dialing number. With AAR and Subnet Trunking, you have a
convenient way to place international calls to frequently-called foreign cities. Such calls route
as far as possible over the private network, and then access the public network. This saves
toll charges and allows you to use your private network as much as possible.
Automatic Route Selection (ARS) selects carriers automatically and routes calls
inexpensively over the public network. When there are one or more long-distance carriers and
Wide Area Telecommunications Service (WATS) provided, Communication Manager selects
the most preferred route for the call. Long-distance carrier-code dialing is not required on
routes selected by the system. You assign long-distance carrier-codes and Communication
Manager translates them. The system inserts codes as needed to guarantee automatic-carrier
selection. ARS can route calls to a variety of types-of-numbers and access a variety of types
of trunk groups.
The Automatic Route Selection (ARS) version of this feature allows users to place calls by
dialing the full public-network numbers without first having to dial a Feature Access Code
(FAC), such as the number "9" to access an outside line. The system recognizes the call as
an ARS call and uses the ARS digit analysis and digit conversion tables to manipulate the
digits to route the call.
The Automatic Alternate Routing (AAR) version of this feature is similar except that the call is
routed as an AAR call and therefore uses the AAR digit analysis and digit conversion tables.
Communication Manager supports overlap sending for AAR and ARS calls that are routed
over ISDN-PRI trunk groups. ISDN-PRI call-address information is sent one digit at a time
instead of in one block. In countries with complex public-network numbering plans, this allows
for a significant decrease in call setup time. When overlap receiving is enabled, this is
especially significant for tandem calls.
Allows AAR and ARS to be partitioned into 8 user groups within a single system and provides
individual routing treatment for each of these user groups.
User groups share the same Partition Group Number, which indicates the choice of routing
tables that are used on a particular call. Each Class of Restriction (COR) is assigned a
specific Partition Group Number or Time of Day specification. Different classes of
restriction may be assigned the same Partition Group Number.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Call Routing Features
Call Routing
Generalized route
selection
Look-ahead routing
Node number routing
Time of day routing
Multiple location
support
Multiple location
support for network
regions
Traveling class marks
Answer detection
Answer supervision
by time-out
Call-classifier board
Network answer
supervision
Provides voice and data call-routing capabilities. You use it to select not only the least-cost
routing, but also optimal routing over the appropriate facilities. It enhances AAR and ARS by
providing additional parameters in the routing decision and maximizing the chance of using
the right facility to route the call. Also, if an endpoint incompatibility exists, it provides a
conversion resource (such as a modem from a modem pool) to attempt to match the right
facility with the right endpoint.
Provides an efficient way to use trunking facilities. It allows you to continue to try to reroute an
outgoing ISDN-PRI call that is not completing. When Communication Manager receives a
cause value that indicates congestion, Look-Ahead Routing tells the system what to do next.
For each routing preference, you can indicate if the next routing-preference should be
attempted or if the current routing-preference should be attempted again.
Allows you to specify the route pattern associated with each node in a private network. It is a
required capability for Extension Number Portability and is used in conjunction with Automatic
Route Selection, AAR and ARS Partitioning, Private Networking, and Uniform Dial Plan.
Uniform Dial Plan extensions can be routed to a specified node using its associated pattern.
Node Number Routing allows a Uniform Dial Plan route pattern based on node numbers or
on location codes. On the AAR and ARS Digit Analysis Tables, you also can specify a Node
Number instead of a Route Pattern.
Provides the most economical routing of ARS and AAR calls. This routing is based on the
time of day and day of the week that each call is made. Up to 8 TOD routing plans may be
administered, each scheduled to change up to 6 times a day for each day in the week. This
allows you to take advantage of lower calling rates during specific times of the day and week.
In addition, companies with locations in different time zones can use different locations that
have lower rates at different times of the day or week. This feature is also used to change
patterns during the times an office is closed in order to reduce or eliminate unauthorized calls.
Multiple Location Support enables local user time, local ARS Public Analysis Tables for local
trunking, automatic Daylight Savings Time, and enhances shared resource algorithms (touch
tone receivers) when Remote Expansion Port Networks (EPNs), ATM Port Networks, and
Avaya Media Gateways are remoted off of a central server at a different location.
See Multiple location support for network regions.
Traveling Class Marks are a mechanism for passing the facility restriction level of a caller from
one Electronic Tandem Network switch to another. Traveling Class Marks allow privilege
checking to be passed across switches through the Electronic Tandem Network.
For purposes of Call-Detail Recording (CDR), it is important to know when the called party
answers a call. Communication Manager provides three ways to determine whether the called
party has answered an outgoing call.
You set a timer for each trunk group. If the caller is off-hook when the timer expires,
Communication Manager assumes that the call has been answered. This is the least accurate
method. Calls that are shorter than the timer duration do not generate call records, and calls
that ring for a long time produce call records whether they are answered or not.
A call-classifier board detects tones and voice-frequency signals on the line and determines
whether a call has been answered.
The Central Office (CO) sends back a signal to indicate that the far end has answered. If a
call has traveled over a private network before reaching the CO, the signal is transmitted back
over the private network to the originating system. This method is extremely accurate, but is
not available in the United States over CO, FX, or WATS trunks.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Reliability and Survivability Features
Reliability and Survivability
Alternate gatekeeper
Auto fallback to
primary for H.248
gateways
The alternate gatekeeper enhancement can provide survivability between Avaya
Communication Manager and IP communications devices such as IP Telephones and IP
Softphone. This is accomplished by providing alternate gatekeepers (CLAN) in the event of
network or gatekeeper failure and by load balancing endpoint traffic among multiple
gatekeepers. It is important to recognize that calls will drop during that interval while the
communication is re-established to the switch.
This feature automatically returns a fragmented network, where a number of H.248 media
gateways are being serviced by one or more Local Survivable Processors (LSP), to the
primary media server. This feature is targeted to H.248 media gateways only.
This feature allows the administrator to define any of the following rules for migration:
•
Whether or not the media gateways, serviced by LSPs, should automatically migrate
to the primary media gateway.
•
Whether or not the media gateway should migrate immediately when possible,
regardless of active call count.
•
Whether or not the media gateway should only migrate if the active call count is 0.
•
Whether or not the media gateway should only be allowed to migrate within a
window of opportunity, by providing day of the week and time intervals per day. This
option does not take call count into consideration.
•
Connection
preserving
failover/failback for
H.248 media gateways
Connection
preserving upgrades
for duplex servers
Whether or not the media gateway should be migrated within a window of
opportunity by providing day of the week and time of day, or immediately if the call
count reaches 0. Both rules are active at the same time.
Internally, the primary call controller gives priority to registration requests from those media
gateways that are currently not being serviced by an LSP. This priority is not administrable.
The Connection Preserving Migration (CPM) feature preserves existing bearer (voice)
connections while an H.248 media gateway migrates from one Communication Manager
server to another. Migration might be caused by a network or server failure.
Only stable calls are preserved. Call that are not preserved are:
•
Unstable calls. An unstable is any call where the call talk path between parties has
not been established, or is not currently established. Some examples are: Calls with
dial tone, Calls in dialing stage, Calls in ringing stage, Calls listening to
announcements, Calls listening to music, Calls on hold (soft, hard), Calls in ACD
queues, Calls in vector processing
•
IP trunks, both SIP and H.323
•
ISDN-BRI telephones
• ISDN-BRI trunks
Users on connection-preserved calls cannot use such features as Hold, Conference, or
Transfer.
The connection preserving upgrades for duplex servers feature provides connection
preservation on upgrades of duplex servers for:
•
connections involving IP telephones
•
connections involving TDM connections on port networks
•
connections on H.248 gateways
• IP connections between port networks and media gateways
Stable calls are preserved. Unstable calls are dropped.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Reliability and Survivability Features
Reliability and Survivability
Enterprise Survivable
Servers
The Enterprise Survivable Server (ESS) provide survivability by allowing backup servers to be
placed in various locations in the customer network. The backup servers supply service to
port networks in the case where the S8500-series media server, or the S8700-series media
server pair fails, or connectivity to the main server or server pair is lost.
In an ESS environment, there can only be one main server, either one S8500-series media
server, or one pair of S8700-series media servers. If the main server is an S8500-series
media server, all ESSs in the configuration must also be S8500-series media servers. During
normal operation, the main server communicates with and controls all the port networks. The
main server contains a license file that identifies the server as the main server and activates
the ESS functionality
Local Survivable
Processor
A Local Survivable Processor (LSP) is an Internal Call Controller (ICC) with an integral G700
Media Gateway, in which the ICC is administered to behave as a spare processor rather than
as the main processor. The standby Avaya S8700 Media Server runs in duplex mode with the
main server ready to take control in the event of a outage with no loss of communication.
An LSP is a configuration used to provide redundancy of the Avaya call processing
application. Usually, a media module serves as the ICC for the system, but it can also serve
as a redundant processor for call processing. In the LSP configuration, the processor serves
as an alternate controller/gatekeeper for IP entities, such as IP telephones and media
gateways. These IP entities use the LSP when they lose connectivity to their primary
controller.
In the event that the communication link is broken between the remote Avaya G700 Media
Gateway and the primary call controller (either an Avaya S8300 Server or an Avaya S8700
Server), the LSP provides service for the Avaya IP telephones and Avaya G700 Media
Gateways that were controlled by the primary call controller.
How the Avaya G700 Gateways and IP endpoints change control from the primary to the LSP
is driven by the endpoints themselves, using a list of call controllers. During initialization, each
IP endpoint and Avaya G700 Gateway receives a list of call controllers. The IP endpoints ask
each call controller in the list for service until one responds with a positive reply. If the link to
that call controller fails at some later time, the endpoint will try to receive service from the
other call controllers in the list, including the LSP.
The LSP provides service to all Avaya G700 Gateways and IP endpoints that register with it.
When the primary call controller is prepared to provide service, the LSP is reset. This informs
the IP endpoints to try their call controller list again, and returns to the primary call controller
for service.
The LSP provides redundancy in a variety of configurations, and can be located anywhere in
a network of Avaya G700 Gateways.
For LSP capacities, refer to the capacities table.
The number of Enterprise Survivable Servers (ESS) that you can administer in one ESS
configuration is increased to 63.
Enterprise Survivable
Server increase
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Reliability and Survivability Features
Reliability and Survivability
Automatic upgrade
tool of server/LSP
software and license
Multiple network
regions per CLAN
Power failure transfer
Survivable Remote
EPN
WAN spare processor
Dial backup over
external ISDN modem
This feature adds the following functionality to the Web page upgrade tool:
•
Distribution of license files from the server on which the upgrade tool is running to
the LSPs that need them
•
Display of SID/MID on the query results
•
Support for the G350 gateways as an upgrade target
•
Upgrade of the standby server
•
Upgrade of the sever on which the upgrade tool is running
•
Upgrade of ESS servers
• Support for FTP as well as TFTP as a gateway upgrade protocol
Support for administration of the number of simultaneous FTP/TFTP sessions This feature is
implemented only on Linux servers.
See Multiple network regions per CLAN.
Provides service to and from the local telephone company central office (CO), including wide
area telecommunications system, during a power failure. This allows you to make or answer
important or emergency calls during a power failure. This feature is also called emergency
transfer.
The Survivable Remote Expansion Port Network (SREPN) allows a DEFINITY ECS (R6r or
later) EPN to provide service to the customer when the link to the main processor fails or is
severed or when the processor or CSS fails. When the links to the system are restored and
stable, the logic switch is manually reset and the EPN is reconnected to the links from the
switch. There are both command and manual resets. The resets can be done remotely at the
SAT or manually at the equipment.
The SREPN must be administered separately (not as a duplicated PPN) to function in a
disaster recovery scenario. It does not function as a survivable remote EPN without the
administration (stations, trunks, features) to support its operation. Note: SREPN is not
compatible with ATM port network connectivity (ATM-PNC). If that is the case, see WAN
Spare Processor.
See WAN Spare Processor.
If there is a primary WAN failure, this feature offers a backup means for the control channel
between the branch office and the main site. The gateway attempts to reestablish the control
channel through an alternate route by dialing back to the main site through an external
modem that is connected to the serial port or the USB port. The external modem, connected
to the PSTN, allows dial-up connection to a router or modem at the main site.
This feature is implemented on the G250 and G350 Media Gateways.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Security, Privacy, and Safety Features
Security, Privacy, And Safety
Access security
gateway
Encryption algorithm
for bearer channels
Alternate facility
restriction levels
Alternate operations
support system alarm
number
Privacy -attendant
lockout
Authorization codes 13 digits
Call restrictions
System Administrator
Access security gateway is an authentication interface used to secure the system
administration and maintenance ports and/or logins on the system. Access security gateway
employs a challenge/response protocol to confirm the validity of a user and reduce the
opportunity for unauthorized access.
Successful authentication is accomplished when the feature communicates with a compatible
key. The challenge/response negotiation is initiated once an RS-232 session is established
and a valid system login ID has been supplied by a user. The authentication transaction
consists of a challenge, issued by the system and based on the login ID supplied by the user,
followed by receipt of the expected response, which is supplied by the user.
Communication Manager supports the Advanced Encryption Standard (AES) format of signal
encryption for IP telephony. This encryption algorithm is in addition to the Avaya proprietary
encryption protocol, the Avaya Encryption Algorithm (AEA).
AES encryption is a cryptographic algorithm developed by the U.S. Government to protect
unclassified information. Communication Manager uses AES with 128 bit keys in counter
mode (AES-128-CTR).
Administration is supported to select a combination of no encryption, AEA encryption, and/or
AES encryption on a per codec set basis.
This feature allows Communication Manager to adjust facility restriction levels or authorization
codes for lines or trunks. Each line or trunk is normally assigned a facility restriction level.
With this feature, alternate facility restriction levels are also assigned. Attendants can change
to the alternates, thus changing access to lines and trunks.
You might want to use this feature to disable most long-distance calling at night, for example,
to prevent unauthorized staff from making long-distance calls.
This feature allows you to establish a second number for Communication Manager to call
when an alarm event occurs. This feature is useful for alerting a second support organization,
such as INADS or OneVision.
See Attendant lockout -privacy.
Authorization codes extend calling-privilege control and enhance security for remote-access
callers. Authorization codes can be up to 13 digits in length.
Avaya site administration authorization codes may be used to:
•
Override facility restriction levels assigned to originating stations or trunks
•
Restrict individual incoming tie trunks and remote-access trunks from accessing
outgoing trunks
•
Track CDR calls for cost-allocation purposes
• Provide additional security control
By dialing an access code, administrators and attendants have the ability to restrict users
from making or receiving certain types of calls. There are five restrictions:
•
Outward. User cannot place external calls.
•
Station-to-station. User cannot place or receive internal calls.
•
Termination. User cannot receive any calls (except priority calls).
•
Toll. User cannot place toll calls but can place local calls.
•
Total. User can neither place nor receive any calls.
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Avaya Communication Manager Feature Overview
Security, Privacy, and Safety Features
Security, Privacy, And Safety
Class of Restriction
Block collect call
Customer-provided
equipment alarm
Data privacy
Data restriction
Facility restriction
levels and traveling
class marks
H.248 link encryption
Malicious call trace
Malicious call trace
logging
Media encryption
Defines many different classes of call origination and termination privileges. Communication
Manager may have no restrictions, only a single COR, or may have as many classes of
restrictions as necessary to effect the desired restrictions. Many different types of classes of
restriction can be assigned to many types of facilities on the switch. For example, you can use
a calling-party COR to prevent callers from accessing the public network.
See Block collect call.
Provides you with an indication that a system alarm has occurred and that the system has
attempted to contact a service organization. A device that you provide, such a lamp or a bell,
is used to indicate the alarm situation. You can administer the level of alarm about which you
want to be notified.
See Data privacy.
See Data restriction.
Allows certain calls to specific users, while denying the same calls to other users. For
example, certain users may be allowed to use Central Office (CO) trunks to other corporate
locations while other users may be restricted to less expensive private-network lines. You can
administer up to eight levels of restriction for users of AAR and ARS.
To provide privacy for media streams carried over IP networks, the H.248 signaling channel
between the media server (media gateway controller) and the media gateways is encrypted.
This signaling channel is used to distribute the media session keys to the media gateways,
and may carry user-dialed authorization codes and passwords.
This feature protects our customer investments by encrypting the signaling channel between
the H.248 gateway and server. This feature also protects the media encryption key, PINs, and
account codes between the media gateway and the media gateway controller.
Encryption of the H.248 link to any given media gateway may be enabled or disabled through
the Media Gateway screen. However, the encryption protocol cannot be disabled.
Allows you to trace malicious calls. You define a group of terminal users who can notify others
in the group when they receive a malicious call. These users can then retrieve information
related to the call. Using this information, you can identify the malicious call source or provide
information to personnel at an adjacent system to complete the trace. It also allows you to
record the malicious call, as well as trace a malicious call over ETSI PRI.
Malicious call trace logging allows a PC to receive information from Communication Manager
to log malicious calls.
Media Encryption is the encryption of the audio (voice) portion of a Voice Over IP (VoIP) call.
Media Encryption can be used to provide enhanced privacy for VoIP communications that
involve exchange of sensitive information. Media Encryption is provided between Avaya
media gateways and media servers.
Digitally encrypting the audio (voice) portion of a VoIP call can reduce the risk of electronic
eavesdropping. IP packet monitors, sometimes called sniffers, are to VoIP calls what wiretaps
are to circuit-switched (TDM) calls. One exception is that an IP packet monitor can watch for
and capture unencrypted IP packets, and can play back the conversation in real-time or store
it for later playback.
Communication Manager encrypts IP packets before they traverse the IP network. An
encrypted conversation sounds like white noise or static when played through an IP monitor.
End users do not know that a call is encrypted because there are:
•
No visual or audible indicators to indicate that the call is encrypted.
•
No appreciable voice quality differences between encrypted calls and non-encrypted
calls.
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Avaya Communication Manager Feature Overview
Security, Privacy, and Safety Features
Security, Privacy, And Safety
Restriction -controlled
Secure shell and
secure FTP
Allows an attendant or telephone user, with console permission, to activate and deactivate for
an individual telephone or a group of telephones, the following restrictions:
outward
•
total
•
station-to-station
• termination restrictions
The Telnet protocol allows remote access to a network device console that is based on login
and password authentication. Beginning with Communication Manager release 3.0, Secure
Shell (SSH) provides this capability over an encrypted channel. Similarly, Secure FTP (SFTP)
is an encrypted version of the FTP protocol that allows remote file transfers. SSH/SFTP
provides a secure alternative for file transfer of firmware download files and voice
announcements, as well as secure remote server access.
The Secure Shell (SSH) and Secure FTP (SFTP) capabilities are highly-secure methods for
remote access. Administration for this capability also allows disabling Telnet when it is not
needed, creating a more secure system.
The enable filexfer command enables SSH and SFTP for both the TN799DP Control LAN
(CLAN) circuit pack, and the TN2501AP Voice Announcement over LAN (VAL) circuit pack.
The Telnet, FTP, SSH, and SFTP enabling capabilities on the TN2312A/BP IP Server
Interface (IPSI) circuit pack continue to be handled through the Communication Manager web
interface and the Communication Manager Linux bash shell.
SSH/SFTP functionality does not require a separate Avaya license, nor are there any entries
in the existing Communication Manager license needed.
Applicable platforms or hardware
You can log in remotely to the following platforms or hardware using SSH as a secure
protocol:
•
G350 Media Gateway
•
C350 Multilayer Modular switch
•
S8300, S8500, S8700, or S8710 Media Server command line
•
IBM e-server BladeCenter Type 8677 command line
•
Communication Manager System Administration Terminal (SAT) interface on a
media server using port 5022.
•
Note: The client device for remote login must also be enabled and configured for
SSH. Refer to your client PC documentation for instructions on the proper
commands for SSH.
•
Secure Shell (SSH) and/or Secure FTP (SFTP) remote access protocols are
provided on these circuit packs:
•
TN799DP (CLAN)
•
TN2501AP (VAL)
•
TN2312AP/BP (IPSI)
•
TN2602AP (Crossfire)
•
SAT commands enable S/FTP sessions through login/password authentication on
the CLAN and VAL circuit packs and SSH/Telnet on the Crossfire circuit pack. The
Maintenance Web Interface and a Communication Manager command line enable
the IPSI session. Unencrypted Telnet and FTP capabilities are enabled on these
circuit packs.
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Avaya Communication Manager Feature Overview
Security, Privacy, and Safety Features
Security, Privacy, And Safety
Security of IP
telephone config files
Security of IP
telephone
registration/H.323
signaling channel
This feature supports the inclusion of a digital certificate and the use of TLS to allow an IP
telephone to authenticate the server for the download of configuration files. This enables IP
telephones to ensure that configuration parameters come only from an authenticated source.
Configuration files that are delivered through this mechanism can deliver message digest
values for the authentication of software code files delivered through a non-secure
connection.
The Security of IP telephone registration/H.323 signaling channel feature provides a secure
mechanism for an H.323 IP endpoint and a Communication Manager gatekeeper to mutually
authenticate each other. The IP endpoint and the Communication Manager gatekeeper
authenticate each other by implicitly showing that each knows the assigned PIN.
The system uses the procedures of H.235.5, formerly published as H.235 Annex H, Security
Profile 1 to accomplish this authentication. The system also used H.235.5 to negotiate a
strong shared secret using the Encrypted Key Exchange (EKE) method.
An authentication key, derived from the master key using a one-way function, is used to
authenticate the contents of the messages. The H.323 endpoint and a Communication
Manager gatekeeper exchange this authentication key during IP registration, admission, and
status (RAS) and during call signaling.
An encryption key, derived from the master key using a one-way function, is used to encrypt
private information that is carried within these messages. Two examples are media encryption
keys and proprietary signaling elements. The proprietary signaling elements carry display
information and dialed digits.
If one or the other parties does not possess the correct PIN, the computed shared secrets are
different. As a result, RAS message authentication fails and the parties refuse to
communicate with each other.
With the Security of IP Telephone Registration/H.323 Signaling Channel feature, the IP
endpoint and the Communication Manager gatekeeper:
•
Authenticate each other
•
Negotiate a strong shared secret
•
Authenticate each message that is sent or received
•
Digitally sign all RAS and call signaling messages
•
Encrypt selected elements of RAS and call signaling messages, such as:
media session keys
proprietary elements
You can also use H.235.5 procedures and security mechanisms for IP trunking by
administering the appropriate Signaling Group screen.
With this feature, the quality of the communication includes:
•
Privacy for selected elements of call signaling, including media session encryption
keys and dialed digits.
•
Security of past or future communications, even if one session is penetrated by an
attacker with knowledge of that session’s keys. This is known as “perfect forward
secrecy.”
•
Efficient reuse of the negotiated strong secret, identified by a unique session ID, to
derive strong keys for new signaling links between Communication Manager and
endpoints, or between other Communication Manager servers, such as IP trunks.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Security, Privacy, and Safety Features
Security, Privacy, And Safety
Security Violation
Notification
Signaling encryption
for SIP trunks
Station security codes
Tripwire security
Backup alerting
Barrier codes
Per call CPN
restriction
Per line CPN
restriction
Security Violation Notification (SVN) allows you to set security-related parameters and to
receive notification when the limits that you have established are violated. You can run reports
related to both valid and invalid access attempts. You can also disable a login ID or remote
access authorization that is associated with a security violation.
Signaling encryption for SIP trunks protects customer investments by encrypting the voice
channel over SIP trunks.
To provide additional security around the customer options the "init" login has been provided
with additional security for the purpose of establishing an authentication procedure for
attempts to remotely log into the system.
Tripwire is a security program provided on all Linux-based media servers. The list of files that
Tripwire monitors needs to be determined during design when all administration and
configuration files have been identified.
If there are any detected security violations, Tripwire reports its findings through the security
log. These events generate an alarm. Note: Tripwire normally reports security violations
through e-mail. However, by reporting events to the security log, security violations can be
immediately acted upon.
End user
Notifies backup attendants that the primary attendant cannot pick up a call. It provides both
audible and visual alerting to backup stations when the attendant queue reaches its queue
warning level. When the queue drops below the queue warning level, alerting stops. Audible
alerting also occurs when the attendant console is in night mode, regardless of the attendant
queue size.
A barrier code is a security code that is used with remote access to prevent unauthorized
access to your system. To increase your system security, use a 7-digit barrier code with
remote access barrier code aging. A barrier code automatically expires if an expiration date or
number of accesses has exceeded the limits you set. If both a time interval and access limits
are administered for a barrier code, the barrier code expires when one of the conditions is
satisfied. Note: Barrier codes are not tracked by call detail recording (CDR). Barrier codes are
incoming access codes, whereas, authorization codes are primarily outgoing access codes.
Calling/Connected Party Number restriction
Users may indicate calling number privacy information. For ISDN calls, the CPN presentation
indicator is encoded accordingly. For non-ISDN calls going to a public network that supports
the CPN restriction feature, the network specific feature activation code gets passed to the
network for interpretation and activation of the desired feature.
If per call CPN restriction is activated for an outgoing call, it will override any per line CPN
restriction administration for the calling station, and will override any ISDN trunk group
administration for sending calling number.
Users may block the calling party number when originating calls. For ISDN calls, the CPN
presentation indicator is encoded accordingly. For non-ISDN calls, going to a public network
that supports the CPN restriction feature, the network specific feature activation code gets
passed to the network for interpretation and activation.
If per line CPN restriction is administered for a station, it will override any ISDN trunk group
administration for sending calling party number.
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Avaya Communication Manager Feature Overview
Security, Privacy, and Safety Features
Security, Privacy, And Safety
Crisis alerts to a
digital numeric pager
Crisis alerts to a
digital station
Crisis alerts to an
attendant console
Emergency access to
the attendant
E911 CAMA trunk
group
Crisis alert can also send notification of an emergency call to a digital pager. In this case, it
sends a message of 7-digits to 22-digits to the pager and displays a crisis alert code, an
extension and room number, and a main number (if one is entered). The person paged thus
knows the origin of the emergency call and can direct emergency-service response to the
appropriate location.
To use crisis alert with a digital pager, the system is administered so that at least one digital
set has a CRSS-ALRT button and the Alert Pager field is set to y. Any station with a CRSSALRT button and a pager receives the correct alert.
Crisis alert uses both audible and visual alerting to notify administered digital display stations
when an emergency call is made. Audible alerting sounds like an ambulance siren. Visual
alerting flashes the CRSS-ALRT button lamp and displays the name and extension, or room,
of the caller. The crisis alert display of the origin of the emergency call enables the attendant
or other user to direct emergency-service response to the caller.
When crisis alerting is active, the station is placed in position-busy mode so that other
incoming calls can not interfere with the emergency call notification. The station can still
originate calls to allow notification of other personnel.
If an emergency call is made while another crisis alert is still active, the incoming call will be
placed in the queue. If the system is administered so that all users must respond, then every
user must respond to every call, in which case the calls are not necessarily queued in the
order in which they were made. If the system is administered so that only one user must
respond, the first crisis alert remains active at the telephone where it was acknowledged.
Subsequent calls are queued to the next available station in the order in which they were
made.
Crisis alert uses both audible and visual alerting to notify attendant consoles when an
emergency call is made. Audible alerting sounds like an ambulance siren. Visual alerting
flashes the CRSS-ALRT button lamp and displays the name and extension, or room, of the
caller. The crisis alert display of the origin of the emergency call enables the attendant or
other user to direct emergency-service response to the caller. Though often used in the
hospitality industry, it can be set up to work with any standard attendant console.
When crisis alerting is active, the console is placed in position-busy mode so that other
incoming calls can not interfere with the emergency call notification. The console can still
originate calls to allow notification of other personnel. Once a crisis alert call has arrived at a
console, the console user must press the position-busy button to unbusy the console, and
press the crisis-alert button to deactivate audible and visual alerting.
If an emergency call is made while another crisis alert is still active, the incoming call will be
placed in the queue. If the system is administered so that all users must respond, then every
user must respond to every call, in which case the calls are not necessarily queued in the
order in which they were made. If the system is administered so that only one user must
respond, the first crisis alert remains active at the telephone where it was acknowledged.
Subsequent calls are queued to the next available station in the order in which they were
made.
Provides for emergency calls to be placed to an attendant. These calls can be placed
automatically by the system or can be dialed by system users. Emergency access calls can
receive priority handling by the attendant.
See E911 CAMA trunk group.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
Security, Privacy, and Safety Features
Security, Privacy, And Safety
Privacy -auto
exclusion
Privacy -manual
exclusion
Restriction -controlled
Station lock
When the class of service (COS) is set for the automatic exclusion option, the feature is
activated when you take your telephone off-hook. The feature can be deactivated when you
push the exclusion button before dialing a call or during a call. An excluded call that is on hold
can be taken off hold by any telephone that has a bridged appearance of the telephone that
put the call on hold.
Allows multi-appearance telephone users to keep other users with appearances of the same
extension number from bridging onto an existing call. Exclusion is activated by pressing the
exclusion button on a per-call basis.
See Restriction -controlled.
Station lock allows users to lock their telephones to prevent unauthorized outgoing calls.
Users can block outgoing calls and still receive incoming calls. This feature is activated by
pressing a telephone button or dialing a feature access code (FAC).
Station lock allows users to block all outgoing calls, except for emergency calls, on all
telephones, unless the telephone is pre-administered. An example of a pre-administered
telephone is a telephone that is administered to block all outgoing calls except for emergency
calls. Telephones can be remotely locked and unlocked.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
System Management Features
System Management
Avaya Communication Manager system management provides the administrator powerful tools to maintain their
communication solutions and to drive down the total cost of ownership.
Administration Without
See Administration Without Hardware.
Hardware
Alternate facility
This feature allows Communication Manager to adjust facility restriction levels or authorization
restriction levels
codes for lines or trunks. Each line or trunk is normally assigned a facility restriction level.
With this feature, alternate facility restriction levels are also assigned. Attendants can change
to the alternates, thus changing access to lines and trunks. You might want to use this feature
to disable most long-distance calling at night, for example, to prevent unauthorized staff from
making long-distance calls.
Announcements
Use the Announcements feature to administer announcements that play for callers to your
business. For example, you can inform callers that the call cannot be completed as dialed, the
call is in a queue, or that all lines are busy. An announcement is often used in conjunction
with music. Announcements can be integrated or external.
•
Integrated announcements reside on a circuit pack in the carrier.
•
Authorization codes -13
digits
Automatic circuit
assurance
Automatic transmission
measurement system
Barrier codes
Bulletin board
Busy verification of
telephones and trunks
External announcements are stored on an adjunct, and are played back from the
adjunct equipment.
See Authorization codes -13 digits.
Assists in identifying possible trunk problems. Communication Manager maintains a record of
the performance of individual trunks and automatically calls a designated user when a
possible failure is detected. This feature provides better service through early detection of
faulty trunks and consequently reduces out-of-service time.
Measures voice and data trunk facilities for satisfactory transmission performance. The
measurement report contains data on trunk signal loss, noise, signaling return loss, and echo
return loss. Acceptable performance, the scheduling of tests, and report contents are
administrable.
See Barrier codes.
Provides a place on the switch where you can post information and receive messages from
other switch users, including Avaya personnel. Anyone with appropriate permissions can use
the bulletin board for everyday messages. In addition, Avaya personnel can leave high-priority
messages that are displayed on the first ten lines of the bulletin board.
Allows attendants and users of multi-appearance telephones to make test calls to trunks,
telephones, and hunt groups to check the status of an apparently busy resource. With this
feature, an attendant or multifunction telephone user can distinguish between a telephone that
is truly busy and one that only appears busy because of some problem. You can also use the
feature to quickly identify faulty trunks.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
System Management Features
System Management
Avaya Communication Manager system management provides the administrator powerful tools to maintain their
communication solutions and to drive down the total cost of ownership.
Call charge information
Provides two ways to know the approximate charge for calls made on outgoing trunks:
•
Advice of Charge, for ISDN trunks - Advice of Charge (AOC) collects charge
information from the public network for each outgoing call. Charge advice is a
number representing the cost of a call; it is recorded as either a charging or currency
unit.
•
Call Detail Recording
Call Detail Recording
display of physical
extension
Call restrictions
Calling party/billing
number
Class of Restriction
Periodic pulse metering, for non-ISDN trunks - Periodic Pulse Metering (PPM)
accumulates pulses transmitted from the public network at periodic intervals during
an outgoing trunk call. At the end of the call, the number of pulses collected is the
basis for determining charges.
Call-charge information helps you to account for the cost of outgoing calls without waiting for
the next bill from your network provider. This is especially important in countries where
telephone bills are not itemized. You can also use this information to let employees know the
cost of their telephone calls, and so encourage them to help manage your company
telecommunications expenses. Note: This feature is not offered by the public network in some
countries, including the United States.
In addition, the pass advice of charge to BRI endpoints feature will transparently pass AOC
information that has been received from PRI networks to WCBRI endpoints.
Records detailed call information on incoming and outgoing calls for the purpose of call
accounting, and sends this call information to a Call Detail Recording (CDR) output device.
You can specify the trunk groups and extensions for which you want records to be kept as
well as the type of information to be recorded. You can keep track of both internal and
external calls. This application contains a wide variety of administrable options and
capabilities.
For Expert Agent Selection (EAS) agent-originated calls, if the Record Agent ID on Outgoing?
field on the CDR System Parameters screen is set to y (the default value), then the agent ID
is used for outgoing calls.
If the Record Agent ID on Outgoing? field on the CDR System Parameters screen is set to n,
the physical extension is used.
By dialing an access code, administrators and attendants have the ability to restrict users
from making or receiving certain types of calls. There are five restrictions:
•
Outward. The user cannot place external calls.
•
Station-to-station. The user cannot place or receive internal calls.
•
Termination. The user cannot receive any calls (except priority calls).
•
Toll. The user cannot place toll calls but can place local calls.
• Total. The user can neither place nor receive any calls.
Allows the system to transmit calling party number/billing number (CPN/BN) information to an
ISDN-PRI trunk group. The CPN is the calling party telephone number. BN is the calling party
billing number. The CPN/BN may contain international country codes. It is used with an
adjunct application.
See Class of Restriction.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
System Management Features
System Management
Avaya Communication Manager system management provides the administrator powerful tools to maintain their
communication solutions and to drive down the total cost of ownership.
Class of Service
Defines whether or not telephone users have permission to access features and functions.
Examples of these features and functions are:
Classless Interdomain
Routing
Concurrent user
sessions
Customer-provided
equipment alarm
Customer telephone
activation
DCS automatic circuit
assurance
External device alarming
•
Automatic callback
•
Call forwarding
•
Data privacy
•
Priority calling
•
Restrict call forwarding off-net
•
Call forward busy/do not answer
•
Extended forwarding and busy/do not answer
•
Personal station access
•
Trunk-to-trunk transfer restriction override
•
Off-hook alert
•
Console permission
• Client room
See Classless Interdomain Routing.
In order to increase the efficiency of administration and maintenance functions, the
Communication Manager accommodates multiple concurrent administration and maintenance
user sessions. Three or more devices (management terminals or operation support systems)
can be connected to the switch to perform administration and/or maintenance tasks
simultaneously.
Communication Manager supports eight concurrent administration and maintenance users.
Five can perform concurrent administration, and three can perform concurrent maintenance.
The eight concurrent sessions can be in any combination of local and remote connections.
See Customer-provided equipment alarm.
Enables customers to install their own telephones, eliminating the need for a service
technician to do the installation. This feature is based on the TTI feature and allows the
customer to associate a physical telephone with a station translations switch.
CTA is a streamlined version of TTI; it has a fixed feature-access code but does not require a
security code. In addition, CTA allows only for "merging" of telephones with station
translations, whereas TTI allows for both "merging" and "unmerging" of telephones with
station translations.
CTA applies only to DCP and analog touch-tone, circuit-switched telephones.
Allows a user or attendant at one node to activate or deactivate automatic circuit assurance
referral calls for the entire DCS network. This transparency allows the referral calls to
originate at a node other than the node that detects the problem.
Allows you to assign analog ports to alarm interfaces for external devices. You can specify a
port location, information to identify the external device, and the alarm level to report when a
contact closure occurs.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
System Management Features
System Management
Avaya Communication Manager system management provides the administrator powerful tools to maintain their
communication solutions and to drive down the total cost of ownership.
Facility busy indication
Allows users of multi-appearance telephones to see which lines, trunk groups, terminating
extension groups, hunt groups, or paging zones (called resources or facilities) are busy.
When the lamp associated with the resource is lit, the resource is busy.
You can store extension numbers, trunk group access codes, and loudspeaker paging access
codes in a facility busy indication button. The facility busy indication button provides direct
access to any of the facilities.
Facility restriction levels Allows certain calls to specific users, while denying the same calls to other users. For
and traveling class
example, certain users may be allowed to use central office (CO) trunks to other corporate
marks
locations while other users may be restricted to less expensive private-network lines. You can
administer up to eight levels of restriction for users of AAR and ARS.
Facility test calls
Allows telephone users to make test calls to access specific trunks, dual tone multifrequency
receivers, time slots, and system tones. The user dials an access code and makes the test
call to make sure the facility is operating properly. Security measures are included to prevent
unauthorized use.
Firmware download
The firmware download feature allows you to download an image from a remote or local
source into the system running Communication Manager, and use that image to reprogram
the application code of a port circuit pack. This feature makes updating firmware more cost
effective. This feature also reduces the expense of servicing the system port circuit packs
because it eliminates the need for a technician to be involved when a board is updated.
Firmware download is achieved using the TN799C CLAN interface. Note: Circuit packs that
can be updated with the firmware download feature have a "P" at the end of their TN number.
Note: This feature is for MCC1 Media Gateways when used with an S8700
Five EPN maximum in
Media Server or DEFINITY® Server R configurations only.
MCC1 Media Gateways
This optional software feature allows customers that require high calling traffic capacities to
have from two to five expansion port networks (EPN) in a single MCC1 Media Gateway. Only
two port networks (PN) are generally available unless a specialized cable was purchased
from Avaya and work-arounds were performed in software administration to make additional
carriers function as EPNs.
When this feature is activated, Communication Manager enables administration of up to five
carriers as EPNs and no custom cables are necessary. This means that the full bandwidth of
the TDM bus is available to each carrier while still enabling the customer to have the footprint
of an MCC1 Media Gateway. This is especially appealing to call centers without IPSI/PNC
duplication, where systems can be quite large and heavily utilized.
The hardware limitation of the MCC1 Media Gateway is five port carriers. All five can be
expansion port carriers, although traffic considerations may dictate some number less than
that which is optimum. For example, a customer may choose to have three EPN carriers and
two standard port carriers.
There is only one maintenance board, which is placed in carrier A. This is the only
maintenance board in the cabinet. Note: Only two PNs are physically supported in S8700
Media Server IPSI-enabled systems when high/critical reliability options are desired. Only two
PNs are physically supported in DEFINITY Server R systems when critical/ATM Network
Duplication reliability is desired.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
System Management Features
System Management
Avaya Communication Manager system management provides the administrator powerful tools to maintain their
communication solutions and to drive down the total cost of ownership.
Information and reports
• Attendant position report - The attendant position report lists the following: -Attendant
usage, -Number of calls answered, -Total time the attendant was available to answer
a new call, -Average holding time on calls answered,
Parsing capabilities
for the history report
•
Blockage study report
•
Call coverage reports - The call coverage report displays measurements of the
distribution of traffic offered to call-coverage groups. Separate reports for all calls
and external calls are supplied.
•
Coverage points report - The coverage points report differs based on whether all
calls or external calls is selected. For each coverage point in the group, the number
of calls offered, abandoned while at that coverage point, and overflowing to the next
coverage point are listed.
•
Display ARP reports
•
Emergency and journal reports - The emergency and journal report is based on
information from all crisis alerts.
•
Hunt group measurements report
•
IP reports
•
Packet error history report - Provides a 24-hour history of important packet level
statistics that indirectly indicate some LAN performance characteristics. The 24-hour
history gives the ability to look back at these measures if the trouble cleared.
•
Port network and link usage report
•
Processor occupancy report - The processor occupancy report provides summary
information on how heavily the processor is loaded.
•
Recent change history report -Allows the system manager to view or print a history
report of the most recent administration and maintenance changes on the switch.
This report may be used fordiagnostic or information purposes.
•
Refresh route reports
•
Summary report - The summary report provides a performance summary of your
system running Communication Manager.
•
Tandem traffic report - The tandem traffic report provides information on facilities that
serve tandem traffic.
•
Tracelog - The Tracelog, among other things, lists: -all IP endpoint registrations -all
IP endpoint unregistrations -all Ethernet interfaces coming into service -all Ethernet
interfaces coming out of service. These events are tagged as a new log type.
•
Traffic reports - Traffic reports show measurements in the format of switch-based
reports for local or remote access, and can be collected for subsequent analysis and
reporting by adjuncts and operation support systems using the operation support
system interface protocol.
• Trunk group detailed measurements
The history report provides details about every data command. You can use parsing options
to limit the data returned in this report. The following parsing options are available.
•
date – Specify the month (MM) or day (MM/DD) for which to display history data.
•
time – Specify the hour (HH) or minute (HH:MM) for which to display history data.
•
login – Specify the login for which you wish to display history data.
©2006 Avaya Inc.
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Avaya Communication Manager Feature Overview
System Management Features
System Management
Avaya Communication Manager system management provides the administrator powerful tools to maintain their
communication solutions and to drive down the total cost of ownership.
IP asynchronous links
Malicious call trace
Malicious call trace
logging
Music-on-hold
Local music-on-hold
Multiple music
sources
Restriction -controlled
Scheduling
Security Violation
Notification
Station security codes
•
action – Specify the command action (the first word of the command string) for which
you wish to display history data. You can view the list of available command actions
by clicking HELP or pressing F5 at the command line.
•
object – Specify the command object for which you wish to display history data.
• qualifier – Specify the command qualifier for which you wish to display history data.
To limit the data displayed in the history report, enter the command list history followed by a
space and the appropriate parser and, if applicable, format. Only the data for the specified
parsers will appear in the report.
You can include multiple parsers, but only a single instance of any parser (for example, you
may parse for date, time, and login, but not for date, time, and two different logins).
See IP asynchronous links.
See Malicious call trace.
See Malicious call trace logging.
Automatically provides music, silence, or tone to a caller. Music lets the caller know that the
connection is still valid.
The music on hold feature is supported on the G700 Media Gateway with Communication
Manager. The music source is connected to a port on the MM711 Analog Media Module.
Local music-on-hold is part of the call center functionality on the S8300 Media Server.
Local music-on-hold allows one music source. To use multiple music sources on a G700
Media Gateway, you must use multiple ports on the MM771 Analog Media Module, one for
each music source.
On an MCC1, SCC1, CMC1, or G600 Media Gateway, this feature allows the customer to
provide multiple distinct music sources for use with the call vectoring features, calls placed on
hole, calls awaiting pickup, and so on. By purchasing the multiple music-on-hold (also called
tenant partitioning) feature, you can have up to 100 music sources.
Many different music options can be administered to accommodate different tenants. See
Tenant partitioning.
Allows an attendant or telephone user, with console permission, to activate and deactivate for
an individual telephone or a group of telephones, the following restrictions:
•
outward
•
total
•
station-to-station
• termination restrictions
Functional scheduling in Communication Manager allows you to specify the time a command
will be executed or to specify that it should be executed on a periodic basis. Only commands
that do not require user interaction after being entered on the command line (such as list,
display, test) can be scheduled.
See Security Violation Notification.
See Station security codes.
©2006 Avaya Inc.
Page 89
Avaya Communication Manager Feature Overview
System Management Features
System Management
Avaya Communication Manager system management provides the administrator powerful tools to maintain their
communication solutions and to drive down the total cost of ownership.
Tenant partitioning
Allows partitioning of the system in order to lease the system services and features to multiple
tenants. This provides attractive services and revenue for "virtual" landlords. It provides the
robust features of a large system at affordable rates to small business tenants.
Communication Manager supports up to 100 partitions and 27 attendant groups.
Multiple attendant groups can be assigned to each partition. Stations, hunt groups, and other
endpoints assigned to a Class of Service (COS) can be partitioned. Network routing pattern
preferences also support the assigned tenant partitioning. Tenant partitioning also allows you
to assign a unique music source for each tenant partition for customers who are put on hold.
Terminal Translation
See Terminal Translation Initialization.
Initialization
Time of day clock
Customers need accurate and common time of day time source across multiple switches in a
synchronization
network. This is especially important when customers are using a central Avaya Call
through a LAN source Management System (CMS) to report events coming from multiple servers running
Communication Manager.
The time of day clock synchronization through a LAN source feature is implemented on two
different platforms:
•
Linux platforms
UNIX platforms
Trunk group circuits
Variable length ping
Variable Length
Subnet Mask
Avaya Integrated
Management
ATM WAN Spare
Processor Manager
Avaya Communication
Manager configuration
manager
Linux
• UNIX
Communication Manager that is running on Linux-based media servers synchronizes time
directly from a LAN source.
Communication Manager running on DEFINITY servers which use an Oryx/Pecos operating
system (proprietary UNIX-based OS) receives a command from Avaya site administration to
adjust the time. Avaya site administration is synchronized to the LAN PC’s clock.
Trunks provide the communications links between Communication Manager and other
switches, including central office switches and other premises switches. Trunks that perform
the same function are grouped together and administered as trunk groups. Trunks interface
with Communication Manager through port circuit packs.
See Variable length ping.
See Variable Length Subnet Mask.
Avaya Integrated Management is a systems management software suite that contains
applications to manage a converged voice and data network. The applications include:
•
network management
•
fault management
•
performance management
•
configuration management
•
directory management
• policy management functionality
See ATM WAN Spare Processor Manager.
Avaya configuration manager provides centralized management of distributed network and
campus environments, using a single point of entry and graphical Web-based interface for
configuration and administration of multiple Avaya media servers.
©2006 Avaya Inc.
Page 90
Avaya Communication Manager Feature Overview
System Management Features
System Management
Avaya Communication Manager system management provides the administrator powerful tools to maintain their
communication solutions and to drive down the total cost of ownership.
Avaya Communication Communication Manager fault/performance manager integrates with Avaya multiservice
Manager
network manager to provide a system view of your converged network. Fault manager
fault/performance
displays a hierarchical view of devices and their status, allowing you to view and isolate
manager
alarms and errors. Performance manager provides a comprehensive set of performance
re