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ANNEXURE-3
Specification of SERVER items no-2 of Schedule-B
1. Communication System Architecture:
1a The IP telephony system should on a IP technology (IP at core) server and gateway type
software solution based communication system. The both Server and gateways can be
geographically dispersed irrespective of whether hosted on LAN or WAN.PCM-TDM based
telephone System (card based processor) shall not be accepted.
1b Built on Server-Gateway architecture to connect to PSTN networks.
1c Call server application should be hardware independent for its deployment on customer’s
choice server..
1d Server BHCC should not be less than 300K
1e It should have support of raid 1 and raid 5 functionalities.
1f Communication Server should be scalable, distributable and the operating system shall be
LINUX based. System shall employ IP at its core with IP switching technology. The Server
should have VOIP and Voice mail server at its core
1g The system should offer maximum availability, with the switchover of call control
processing functions to an alternate or redundant processor in the event of significant
fault. The redundancy scheme should conform to the model used in most computer
systems: the complete "mirroring" of the information (both static and dynamic data.) The
switch over between 2 redundant call servers should not interrupt existing and
established communications to include all analog, hard, soft and Video IP Phones. The
complete set of programs and software modules must be duplicated in real time. In case
of failure of the main Server (hardware or software), the Active standby Server
(emergency mirror) must take over the control of communications instantaneously.
1h All critical resource elements (call server, hard disks, data bases, IP interface cards,
Processor, etc. must be redundant and in a hot-standby configuration.
1i In case of failure there should be a Standby server placed in the network to take care of
the call initiation without any delay.
1j Server should support auto sync option for configuration and real time data
1k There should be no delay in progress/initiation of calls after the failure of the main server.
1L Shall support Active/Active redundancy.
1m The communication servers must work in an Active/ Active with Geo-redundancy mode. If
one active server fails, other
active server must be able to take the complete load of the calls automatically (without
any manual intervention) without
Dropping the Established calls.
1n There should be no restriction on number of endpoints being backed up in case of Call
server failure.
1o The redundant server should be separate Hardware not sharing elements like Hard drives,
RAM, etc. to avoid a single point
of Failure
1p Encrypted signalling between Server-Terminals, Server-gateways and SIP Terminal-SIP
terminal
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1u All telephony servers must form a single cluster and should have common database so
that any gateway can be registered to
any server without any problem in case of Network / server failure. Replication of
database should be in real time between
Servers.
1v The Media Gateways, IP Telephones, UC application software comprising of all modules
(Conferencing, call billing, Logging/recording, voice mail, attendant, mobile application
etc.) should be of same Make/ OEM only. Third party Gateways or IP phones will not be
accepted to avoid multiple OEM dependency.
1w The communication system should support 99.99% uptime of internal and external
communications with no single point of failure. It should allow addition of new applications
and new sites with lot of ease.
1x Call processing for all users shall be centralized for common CDR for all the sites, unique
extension numbers across all the sites.
1y It should be feasible to increase the system capacity till 5000 ports without changing the
server hardware or server
software
1z Communication system Should be equipped with built in Unified communication server
1a.1 UC server user capacity should be expandable without changing the server hardware or
server software
1b.1 Redundancy shall be supported for UC server application.
1c.1 Audio Conference Server (ACS) application shall be built in Call server; it shall not
require separate server hardware.
Server should support 125 participants in single conference.
1d.1 Audio conference server should support 500 users.
1e.1 All UC users can publish their status at a time
1f.1 It shall support Built in Voice mail server with 64 voice mail channels
1g.1 All Users of communication system should have access to their Voice Mail boxes
1h.1 Support of 64 simultaneous Voice Recording of calls.
2. Communication Server Capabilities and Capacity:
2.a Active/Active servers in cluster should be expandable up to minimum of 10000 ports. In
any combination (Analog/Digital/CO/PRI/IP etc) Documentary Proof need to Submitted for
this purpose.
2.b The system should be capable of supporting trunks / subscribers of IP, Analog, Digital and
SIP from day one. All necessary
Hardware/software should be provisioned from day one..
2.c The gateway shall be non-blockable i.e. all IP Phones should able to call all type Users
at any point or vice versa. For this provision tenderer must enclose documentary proof
with the tender document.
2.d System shall support following signalling Protocols- (i). ISDN PRI, CEPT,(ii). ISDN BRI,
(iii). E&M (2/4 Wire), (iv). Ring Down, (v). DTMF, (vi).Loop Tie on DTMF, (vii). All the
Common Signalling Standards adopted for ISDN / Non ISDN connections to PSTN, (viii).
SIP
2.e The System should have Voice Logger/Call recording:- The system should have same
OEM integrated 98 port active voice logger, possible to record any conversations ( analog
/ SIP user and analog /PRI / SIP trunk ) on an HDD or USB memory to be directly
connected to the system for a minimum period Voice recording of 30 days. It shall also
be possible to store conversations on a storage device on the IP network. Third party
Voice logger will be rejected. Bidder should consider voice logger in price schedule.
2.f It shall support storage of voice messages in network drive
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2.g Up to 125 party Audio Conference should be supported (any Combination)
2.h The communication server must build up high reliable software architecture running on
Linux operating system on industry grade (COTS) rack sever
2.i The offered communication server should provide communications solutions over IP, no
restriction should evolve in terms of quality of service, reliability and security
2.j The server system should support networking of two or more communication server over
IP infrastructure
2.k Provide open interfaces and standard protocols for current and future applications
2.l Support of session initiation Protocol (SIP) to provide interface connection to ITSP and
service providers
2.m Support 64 port Voice mail server, Voice mail server shall support features like call
queuing, multi-language support, multiple MOH
3. Communication Server Configuration:
IP –server should be commercially of the shelf (COTS) and should not be a
proprietary, must be 19” rack mountable like HP/Dell/IBM/Lenovo and should have
following specification:-
i. Processor: - Intel Xeon E3-1200 v6 or latest
ii. RAM: - 24 GB (min) DDR4.
iii.HDD:- 2 TB or higher – 02 Nos.
iv. Should have RAID controller.
v. Should have DVD R+W (internal or External).
vi. LAN Port Interface:- Should have minimum 4 nos. Gigabit Ethernet
Port internal or external.
vii. Support for high availability clustering & virtualization.
viii. Power Supply: 230V AC power supply.
X .CPU speed 3.10 Ghz or higher
License & software for Server: From OEM. No Third party software/License will be
considered.
4. System Security:-
The system must incorporate advance security features like real time medial
encryption.
System should have facility to disable Telnet and FTP server access to have
maximum security.
Support of SIP over TLS and SRTP without any licenses. It should be IPv6
ready from first day
System SIP trunk must accept traffic from trusted IP source and it must
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support digest authentication for security of SIP traffic.
System should block GUI access after certain unsuccessful attempts of login.
Also it should support Password Ageing.
Operating System used by the communication system must not use or
natively support network resource sharing services such as NFS, samba,
LPR etc.
IP Phones should not support direct, external initiated, connections via HTTP,
telnet, FTP, TFTP or any other protocol as means to prevent distributed
Denial of Service attack exploitation.
IP Phones must support 802.1x (EAP-MD5 or better) for authentication and
access control to the network, this mechanism must allow the user to be
connected to the system once he has passed the authentication process; not
before
System should have options to configure voice VLAN number , allowing for the
separation of voice and data traffic
Administration users connecting directly to the Call Server (console) shall
be authenticated
All management traffic between a remote console/session and the system
must be encrypted. (HTTPS for web sessions etc.)
The management platform must provide Role Based Account Management to
define different levels of administrator access depending on specific function
responsibility.
5. Software Up gradation, Maintenance & Management:
Web based GUI for maintenance, administration and configuration. Dedicated
programming terminal should not be required.
System should support remote configuration Over WAN or any computer in LAN
System should support up gradation with direct GUI upload and there must
minimum breakdown while up gradation of system.
The system can be programmed through Ethernet directly with online GUI
without any external devices or modem.
The system shall have a built-in remote maintenance facility. The system can
be programmed remotely over the internet without any modem required on
the System side
System should maintain logs of all faults occurred.
Provision of notifying system admin by sending email on specific email Id.
Reports of faults/activity should be available through GUI in Printable format
System should have buffer of storing minimum 500 faults/error logs
System Fault logs should be available in online/offline mode
Fault log notification on operator console.
SNMP support for warning messages traps, Errors must be sent to SNMP
server with any SNMP v1, v2 & v3 protocols.
System should have notification of all alarms, logs to be saved in system.
System shall support notification of faults over Email
System must allow simultaneous GUI login from with 3 levels of access
Management platform must provide a single graphical client (Graphical
User Interface (GUI))
Management platform must provide web access allowing the administrator
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to manage the system to use any PC with an internet browser.
Configuration and Programming of services, users, categories and all system
parameters and features. This must provide management in local or remote
environments of a single system or a network. The network manager will be
able to configure all locations users
Fault and error log notification availability on NMS software
Generate reports about faults and errors in PDF format
Access to exchange directory should be available through system GUI.it
should be possible to Export/import Exchange directory in excel, csv format.
Integrated Directory. It shall be possible to provide display equipped voice
terminals with access to system directory on IP phones.
System shall have the provision of automatically identifying and isolating
faulty trunks. This should be done on daily basis automatically or invoked
by System administrator and results output on Maintenance terminal.
It should relieve the user from remembering lengthy procedure and formats
for data changes and shall use simple English commands.
It should have options to record voice help message that can be available
to all users by dialing voice help code.
The administration should be secured by way of password protection. It
should be possible to have different layers of password.
The system management terminal shall be capable of adding/ modifying
hardware/software at main location and remote Location from main location
only
System usage: The usage display should enable users to view the
results of running calls.
Performance/status/information. The software, through real time monitoring
should be able to provide the status of extension and trunks to the EPABX
performance at any time on request basis through screen displays
Fault Detection/Alarm: The software should constantly monitor the exchange
performance and report/generate suitable alarms during any failures to
indicate/localize the faults besides keeping the error-logs for various hardware failure
detected.
6. The system shall have below telephony features:
ARS (Automatic Route Selection)
Auto Attendant
Account Coded Call
Automatic Call Back
Audio Conference
Add-on conference
Internal Call
Trunk Call
Selective Line Access
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Changing Account Code Password
Last Number Redial
Private Pool Entry
Calling from Private Pool
Common Pool Entry
Calling from Common Pool
Common Pool Auto-dial
Changing Password
CLIR / COLR
Call Hold
Call Transfer
Call Back
Camp On
Call Park
Call FORWARDING
Last Number auto-dial
Line Flash
Line Transfer to an Extension
Line Transfer to Extension on Hold
Selective Line Transfer to Extension on Hold
Line Transfer to a Parked Call
Answering the second Call
Selective Call Pick Up
Busy Call Pick Up
Group Call Pick Up
Serial Transfer
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Retrieving a Call Parked through On Own Phone
Retrieving a Call Parked through Another Phone
Retrieving a Parked Line
Busy Lamp Field (BLF)
Bridged line appearance(BLA)
Break In
Intrusion
Intrusion with beep
Malicious Call Identification
Line Drop
Busy Extension Signaling
Line-Line Connection
Hot Line facility
Follow Me
Remote Follow Me
Follow Me No Answer
Follow Me Busy
Extended Call Forward
Password Define
Telephone Lock
Authorization Codes.
Password Dialing
Phone Lock Level
Remote Line Access
Parallel Ringing / Tone on Busy
Reminder Service
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Do Not Disturb
Date Setting
Time Setting
Night Mode
Wake-Up Service
Group Wake-Up Service
Message Waiting Indicator(MWI)
Reminder or wake-up message examples
Direct inward station access (DISA)
The services related to the direct outward system access(DOSA)
Leaving Message to an Extension through the DISA Line
Listening to the Messages through the DISA Line
Direct Inward dialing (DID)
Direct Outward Dialing (DOD)
Day/Night Class of service
Distinctive ringing
Priority Calling
Parallel Ringing feature
Camp on no response/busy
Dynamic locking
One number service
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7 Specification of Mobile UC client for Android/IPhone:
ShallbeinstalledonandroidOS5.1orlater
ShallbeinstalledonIOS7orlater
Comprehensive Call Management ,One-Touch Access to PBX Features
Corporate Directory Integration, Video Calling, Favorites
Presence Sharing and Instant Messaging, Voice Mail Access
Should Support ConversationRecording,Callmanagement–
callhold,transfer,forward,DND and intercom
Multi party audio conferencing, Blind transfer
Should Support Menu options-Call Pickup, Paging, Message wait, Call
Retrieve, Alarm and Reminder ,Dynamic Lock, Dial-In Conference, CLIR,
Room monitoring & Call supervision
Dial by extension
Should Support Call Screening-ACB, Forced Answer, Global Hold,
General Call Park, Call Chaining, IR &Barge-IN
Should Support Multiple call support, One touch transfer, Wi-Fi to cellular
handover
Should Support
Multiplelanguagesupport,Calltoggle,Autocallback,Autoredial,Forcedanswer
Should Support Open a door, Call logs, Missed calls notification, DND override
8 Web based audio & Video collaboration and Conferencing with suitable
hardware from same OEM.
The following features are required for all IP Telephone users who shall use IP
telephone, Desktop (PC/Laptop) & Mobile device with a single number to access the
features on 24 IP users as a single Group & minimum 4 number of groups shall allowed
to work from day 1 These groups shall be manageable. Documentary evidence
regarding this from the OEM’s Technical Manual and/or Datasheet must be submitted
for all the below listed features along with the bid. Also bidder may be asked to
demonstrate the function during technical evaluation of the bid. Bidder shall provide all
necessary hardware/ software require for the feature listed below:
This application should provide a single access point for all the users communication
and collaboration needs. It should converge the call control capabilities of the IP-PBX
platform with contact management, Dynamic Status, and collaboration applications, to
simplify and enhance real-time communications.
Softphone integration with Audio, Web and Video Conferencing:-It should provide
access to collaboration features such as real-time audio, video, and Web conferencing,
whiteboard, desktop and application sharing, and file transfer from the Client. The
bidder should provide minimum 24 conferencing license. Soft phone integration with
Audio, Web and Video Conferencing application should allow users to place a video call
with the click of a button, and create or schedule collaboration sessions.
Public and private chat - Instant messaging shall enable group chat and private chat
side bar interactions to occur between participants without disrupting the overall
meeting. All participants should be able to save a transcript of the instant message
session enabling them to obtain a copy of the discussion.
Desktop & application sharing- During an active session, participants should easily
share their entire desktop, a region of their desktop, or a specific
application/document, allowing for spontaneous content sharing at any point whether
connected via conferencing desktop client or web-based connection.
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Polling- Moderators should easily solicit feedback from the session audience with
targeted polling questions. These can be used at the end of a session to gain feed-back
on the topic just discussed or used throughout the session to maintain participant
focus. The results shall be shared with everyone or kept hidden from view.
Session Border Controller - Should provide a secure communications path for remote
Softphone Client users over Wi-Fi/Internet/3G/4G network to avail the office softphone
services from home or on the road, without the need for a virtual private network
(VPN)
Connection.
It can be possible to Customize each conference with multiple options including entry
tones, recording option, enter conferences mute on or off
It should have facility for Real-time views of conferences and participants with audio
controls, along with transfer or dropping participants
Secure Instant Messaging – Should facilitate secure instant messaging & initiating a
single or multi-party chat at the click of a mouse.
File Sharing/Transfer - Share documents by dragging and dropping files into the chat
session including Files, Images, Audio, and Location details
It should be designed to run across the Internet or a private data network without
requiring any changes to firewalls. Running behind a secure private data network
insures that your web conference content cannot be compromised.
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