Professional Documents
Culture Documents
KazTransCom
KE01-A1-990-KT-T-ST-0501-000
PROJECT:
KASHAGAN DEVELOPMENT
EXPERIMENTAL PROGRAM
CONTRACT DESCRIPTION:
INTEGRATED TELECOMMUNICATION
SYSTEM
TELECOMMUNICATION SYSTEM
TELEPHONE SYSTEM – MAIN CONTROL
BUILDING-TECHNICAL SPECIFICATION
ABSTRACT:
This document details the technical aspects of
TELEPHONE/PABX SYSTEM integrated solution for Main
Control Building
TABLE OF CONTENTS
1 INTRODUCTION............................................................................................................5
2 REFERENCE DOCUMENTS.........................................................................................6
3 CODES, STANDARD AND DEFINITIONS....................................................................7
3.1 Glossary.............................................................................................................................. 7
3.2 Industry Codes and Standards............................................................................................8
3.3 Unit of Measurement........................................................................................................... 9
3.4 Definitions........................................................................................................................... 9
4 TELEPHONE/PABX SYSTEM...........................................................................................11
4.1 Telephone/PABX System network....................................................................................11
4.1.1 General..................................................................................................................................... 11
4.1.2 Telephone/PABX System Architecture..................................................................................12
4.2 Telephone/PABX System components..............................................................................13
4.2.1 Media Gateway........................................................................................................................ 13
4.2.2 Voice Announcement over LAN (VAL)..................................................................................14
4.2.3 TN2312BP IP Server Interface (IPSI)......................................................................................14
4.2.4 TN799DP Control LAN............................................................................................................ 15
4.2.5 TN2214CP DCP digital line (2-wire, 24 ports)........................................................................15
4.2.6 TN793CP analogue line with Caller ID (24 ports)..................................................................15
4.2.7 TN2464CP DS1 interface with echo cancellation, T1/E1......................................................16
4.2.8 TN747B central office trunk (8 ports)....................................................................................17
4.2.9 TN771DP maintenance and test............................................................................................. 17
4.2.10 TN2602AP IP Media Resource 320......................................................................................... 18
4.2.11 Media Server............................................................................................................................ 19
4.2.12 Ethernet switches................................................................................................................... 21
4.2.13 Dial-up modem........................................................................................................................ 21
4.2.14 Voice Mail................................................................................................................................. 21
4.2.14.1 Voice Messaging Features.................................................................................................22
4.2.14.2 Voice Messaging................................................................................................................ 22
4.2.14.3 Call Answer........................................................................................................................ 22
4.2.14.4 Automated Attendant......................................................................................................... 23
4.2.14.5 Bulletin Board..................................................................................................................... 23
4.2.15 Call billing/logging subsystem.............................................................................................. 23
4.2.16 Communication Manager Software.......................................................................................24
4.2.17 Voice Distribution sub-system............................................................................................... 24
4.2.18 Static Transfer Switch............................................................................................................. 24
4.2.19 Fibre Optical Multiplexer FOM-4R..........................................................................................26
4.3 Telephone/PABX System IP configuration........................................................................27
4.3.1 Onshore Main Control Building.............................................................................................27
4.4 Telephone/PABX System E1 configuration.......................................................................29
4.5 Telephone/PABX System power consumption..................................................................30
4.5.1 Onshore Main Control Building active equipment.....................................................................30
1 INTRODUCTION
The document describes the main technical details of the Telephone/PABX System for Main
Control Building and provides the references to the related design/engineering documentation.
2 REFERENCE DOCUMENTS
Kashagan Field Development Project -Experimental Programme Telecommunication
1
Systems ITT: LN-0118-2003-0453
2 2003-0453 Section 6.9 EP Telecomms
3 Section 4 Att 3 - TELECOMMUNICATION SYSTEMS Attachment 3 – Onshore
4 Section 4 Att 4 - TELECOMMUNICATION SYSTEMS Attachment 4 – Offshore
Contract Number 2003-0453 Schedule D Section 4 SUMMARISED SCOPE OF WORK
5
FOR TELECOMMUNICATION SYSTEMS
6 KE01-00-990-KD-T-ST-0003-000 GENERAL TELECOMMUNICATIONS SPECIFICATION
7 KE01-00-990-KT-T-SF-0501-000 TELEPHONE SYSTEM - FUNCTIONAL SPECIFICATION
8 KE01-A1-990-KT-T-TD-0501-000 TELEPHONE SYSTEM - DATA SHEETS
9 KE01-A0-990-KT-T-HB-0501-001 TELEPHONE SYSTEM - ONSHORE BLOCK DIAGRAM
10 KE01-A1-990-KT-T-HT-0503-001 TELEPHONE SYSTEM - TERMINATION SCHEDULE - MAIN CONTROL BUILDING
11 KE01-00-990-KT-T-ZZ-0301-000 DATA NETWORK SYSTEM - TABLE OF OVERALL IP ADDRESS STRUCTURE
12 KE01-A1-990-KT-T-HW-0501-00x TELEPHONE SYSTEM – WIRING DIAGRAM - MAIN CONTROL BUILDING
13 KE01-A1-990-KT-T-LT-0501-000 TELEPHONE SYSTEM – BILL OF MATERIALS - MAIN CONTROL BUILDING
14 KE01-A0-990-KT-T-HB-0501-001 TELEPHONE SYSTEM - BLOCK DIAGRAM – ONSHORE
15 KE01-A1-990-KT-T-HH-0501-001 TELEPHONE SYSTEM - SCHEMATIC DIAGRAM – MAIN CONTROL BUILDING
16 KE01-A1-990-KT-T-DA-0501-00x TELEPHONE SYSTEM - GENERAL ARRANGEMENT DRAWING - MCB
17 KE01-A1-990-KT-T-PR-0501-000 TELEPHONE SYSTEM – FAT PROCEDURE – ONSHORE - MAIN CONTROL BUILDING
18 KE01-A1-990-KT-T-PR-0504-000 TELEPHONE SYSTEM – SAT PROCEDURE – ONSHORE - MAIN CONTROL BUILDING
3.1 Glossary
IP Internet Protocol
PN Port Network
Other abbreviations and expressions when used will be defined within the text.
3.4 Definitions
The COMPANY is the party that initiates the project and ultimately pays for its design and
construction. The Company will generally specify the technical requirements. The Company may
also include an agent or consultant authorised to act for, and on behalf of, the Company.
The CONTRACTOR is the party that carries out all or part of the design, engineering,
procurement, construction, commissioning or management of a project, or operation or
maintenance of a facility. The Company may undertake all or part of the duties of the Contractor.
1+1 - A fully operational spare unit or module dedicated to replace a particular unit or module in
the event of failure of that module.
N+1 - A spare (electrical interface, card or internal unit) which can replace one of N identical
units automatically, by remote control or manually.
Sub-contractor- The party(s) which carry(s) out all or part of the design, procurement,
installation and testing of the system(s) as specified by the Contractor/Supplier.
Central Control Room - The section of a control centre containing the essential equipment
required to operate the plant optimally and safely.
Configuration - The processes of designing and setting up the system using hardware
components, modules and software.
Console - An enclosure housing a group of workstations c/w the VDUs, keyboards and
electronics which will be ergonomically designed to provide a comfortable seated arrangement
for the user to interface with the operator workstation.
Ergonomics (human factors) - The application of human physical and cognitive sciences in
conjunction with the engineering sciences to achieve the optimum human performance and
interaction between a human user and a machine.
Fire & Gas system - A system that incorporates fire & gas sensors/detectors and the logic
solver to provide fire, flammable and toxic gas detection and response in the event of a loss of
containment or other hazardous conditions. Outputs may be executed by the system directly, or
through other systems.
Network Management System (NMS) - a computerised system that communicates with the
remote equipment so that faults in these devices can be diagnosed and resolved remotely.
These devices can also be re-ranged and re-configured from the NMS.
Subsystem - A single or group of modules and/or assemblies providing a specific function which
when interconnected create the System.
System – The Telephone/PABX System as defined and described within this document.
4 TELEPHONE/PABX SYSTEM
4.1 Telephone/PABX System network
General
Main Control Building Telephone exchange will be integrated in the existing Company
Telephone System network. See paragraph 4.4 Telephone/PABX System E1 for the details.
The functionality provided by the Telephone System network is described in the reference
document [7].
The details of the installed equipment in the above-mentioned racks can be found in the General
Arrangements drawings [16].
The voice distribution equipment and IDF is installed in the following locations:
The installed equipment general arrangements can be found in the General Arrangements
drawings [16]
The telephone exchanges require transport media for the following purposes:
-Voice communications
-Management data communications (SNMP, call logging, system access etc.)
The voice communications are ensured over E1 bearers which provide non-compressed PCM
64kbps voice channels. For the E1 configuration refer to the section 4.4. The voice
communications between MCB and other locations will use SDH as transmission network.
The data communications are provided via Data Network System and shall enable the SNMP
system management by NMS, call logging data gathering and administration. Refer to the
section for further details.
(i) Media Gateway (see paragraph 4.2.1 Media Gateway for the details);
(ii) Media Server (see paragraph 4.2.11 Media Server for the details);
(iii) Ethernet switches for system interoperability (see paragraph 4.2.12 Ethernet switch for
the details);
(iv) Voice mail system (see paragraph 4.2.14 Voice Mail for the details);
(v) Call Logging/billing software (see paragraph 4.2.15 Call billing/logging subsystem for the
details);
(ii) MDF, DDF and IDF (see paragraph 4.2.17 Voice Distribution sub-system for the details);
(iii) RJ-45 Patch panels for inter system connectivity and for voice exchange/distribution
panels;
(v) Static Transfer Switch (STS) which ensures 1+1 power feed redundancy for single
power supply equipment (see paragraph 4.2.18 for the details);
(vi) Fibre Optical Multiplexer (see paragraph 4.2.19 Fibre Optical Multiplexer FOM-4R)
The Avaya G650 Media Gateway delivers the scalability, features, and system uptime that
enterprises require for mission-critical telephony applications in campus and large office
environments. The G650 seamlessly integrates traditional circuit-switched and IP-based
telephony networks, providing core telephony gateway services to all endpoints in a mixed
TDM/IP environment, including:
The Avaya G650 Media Gateway allows evolving easily from circuit-based telephony
infrastructures to the next generation of IP infrastructures, including those based on the open
SIP (Session Initiation Protocol) standard.
Avaya solutions are based on a modular architecture of centralized media servers that provide
call processing and control through Avaya Communication Manager and a distributed network of
media gateways. For call control, the G650 Gateway connects to an external S87 30 Media
Server. This enables the design and administration of telephony services for all gateways from
one central location.
For scalability, up to five G650 Media Gateways can be combined to form a high-capacity port
network that may be integrated into existing telephony networks, supporting voice connectivity
over IP and TDM transport. The G650 supports a variety of network availability and transport
configurations, allowing it to be integrated into existing networks to provide a physical integration
path that allows an enterprise to consolidate their voice/data infrastructure onto a common form
factor. For enterprises that demand no compromise TDM/IP telephony support, the G650 Media
Gateway provides a highly scalable solution that combines the full benefits of IP with the
mission-critical availability of traditional TDM telephony.
Networks PN1, PN2, PN3 are connected [12]. Each network is complemented by the TN2312BP
IP Server Interface (IPSI) circuit pack.
The Avaya G650 Media Gateway provides a single 8U high, 14-slot chassis that can be installed
in industry standard EIA-310 19”, ETSI closed racks. Available dual redundant, load-sharing
power supplies with AC/DC inputs enhance system reliability by providing N+1 redundancy and
optional connectivity to back-up power sources. For scalability, up to 5 G650 gateways can be
stacked using a TDM/LAN cable and a built-in connector in the back of the chassis.
The dedicated Voice announcement service is provided using TN2501AP Voice Announcement
Over LAN (VAL) circuit pack which is an integrated announcement circuit pack that:
(vi) 10/100 Mb Ethernet interface, allowing announcement and firmware file portability over
your LAN (FTP server functions).
Voice Announcement over LAN (VAL) requires that announcement files are in the following
*.wav formats:
-CCITT A-Law or CCITT μ-Law (mu-Law) companding format (do not use PCM)
- 8-kHz sample rate
- 8-bit resolution (bits per sample)
-Mono (channels = 1)
In configurations with the S8730 Media Server controlling media gateways, the bearer paths and
the control paths are separate. Control information for port networks (PNs) travels over a LAN
through the Ethernet switch. The control information terminates on the S8730 Media Server at
one end and on a TN2312BP IP Server Interface (IPSI) on the other end. Each IPSI may control
up to five port networks by tunneling control messages over the Center-Stage to PNs that do not
have IPSIs.
Note: IPSIs cannot be placed in a PN that has a Stratum-3 clock interface. Also, IPSIs cannot be
placed in a remote PN that is using a DS1 converter. In configurations that use a dedicated LAN
for the control path, IPSI IP addresses are typically assigned automatically using DHCP service
from the S8730. Also, a dedicated IPSI Ethernet connection to a laptop can be used to assign
static IP addresses or for maintenance. In configurations using the customer’s LAN, only static
addressing is supported.
Telephone systems are interconnected by both tie trunks (for voice communications) and data
links (for control and transparent feature information). Various DS1, IP, and analogue trunk
circuit packs provide the voice-communications interface. For TCP/IP connectivity, the data-link
interface is provided by a TN799DP Control LAN (C-LAN) circuit pack.
The C-LAN handles the data-link signaling information in one of two configurations: Ethernet, or
point-to-point (PPP). The C-LAN circuit pack has one 10/100baseT Ethernet connection and up
to 16 DS0 physical interfaces for PPP connections. C-LAN also extends ISDN capabilities to csi
models by providing packet-bus access.
In the Ethernet configuration, the C-LAN passes the signaling information over a separate
TCP/IP network (via Ethernet switch).
In the PPP configuration, the C-LAN passes the data-link signaling to the DS1 for inclusion in the
same DS1 bit stream as the DCS voice transmissions. For this configuration, install the C-LAN
circuit pack; no other connections are needed. The appropriate DS1 circuit packs must be
installed, if they are not already present.
The CLAN boards are used for dedicated Ethernet connections to TNMS, Billing workstation,
Voice Mail server.
The TN2214 has 24 DCP ports that can connect to 2-wire digital telephones. Such telephones
include 2400- and 6400-series telephones.
The TN793CP is an analogue line, 24-port circuit pack that supports caller ID telephones and
caller ID devices that conform to Bellcore Standard GR-30-CORE, Issue 2, and Bellcore-
compliant signaling using V.23 Frequency Signal Keying (FSK). Each port can support one of
the following:
- FAX
-Loop-start CO port (used for INTUITY AUDIX Messaging)
The TN793CP supports on-premises (in-building) wiring and off-premises wiring with either
DTMF or rotary dialing, but LED or neon message waiting indicators are not supported off-
premises.
The TN793CP supports three ringer loads. A maximum of twelve ports can be rung
simultaneously. To achieve this maximum, the system uses four ports from the set of ports
numbered one through eight, four ports from the set of ports numbered 9 through 16, and four
ports from the set of ports numbered 17 through 24.
The TN793CP circuit pack supports A- and μ-law companding and administrable timers. The
TN793 circuit pack supports queue warning level lights. These lights are associated with the
direct department calling (DDC) and the uniform call distribution (UCD) features, recorded
announcements that are associated with the Intercept Treatment feature, and PagePac paging
system for the Loudspeaker Paging feature.
The TN793CP provides -48 VDC current in the off-hook state. Ringing voltage is -90 VDC. The
TN793CP supports DTMF sending levels that are appropriate for Avaya Interactive Response.
The TN793CP circuit pack’s multinational support is identical to that of the TN2215 circuit pack.
Therefore, the TN793CP allows country-specific transmission selection. The TN793CP is also
impedance and gain selectable for multiple countries.
The following table lists the TN793CP-supported telephones and shows each of their wiring
sizes and ranges.
The analogue lines are used to provide interface PA/GA and Voice Mail systems.
The TN2464CP has echo cancellation circuitry and firmware download capability. The
TN2464CP supports T1 (24-channel) and E1 (32-channel) digital facilities. In ISDN-PRI
applications, the ISDN D channel connects the TN2138 central office trunk (8 ports) by the LAN
bus.
The TN2464CP can be updated using the firmware download feature, which requires use of the
TN799 C-LAN interface.
The TN2464CP is used for inter-PABX connection (see also section 4.4 Telephone/PABX
System E1 configuration).
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. 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.
The TN771DP maintenance test circuit pack performs maintenance functions. These functions
include packet bus reconfiguration. This reconfiguration allows diagnosis and correction of
recoverable packet bus failures before the link access procedure on the D-channel (LAPD) links
fail. LAPD is a link-layer protocol on the ISDN-BRI and ISDN-PRI data link layer (level 2). LAPD
provides data transfer between two devices and error and flow control on multiple logical links.
LAPD swaps spare leads with the malfunctioning leads to recover packet bus failures that
involve up to three malfunctioning leads. Malfunctioning leads, in this case, are 1 or 2 data or
parity leads and one control lead.
Other maintenance functions include ISDN-PRI testing that originates and terminates loopback
tests on ISDN facilities. The testing provides bit and block error rate information that indicates
ISDN facility quality. The TN771DP circuit pack can be updated using the firmware download
feature, which requires use of the TN799 C-LAN circuit pack interface.
The TN2602AP IP Media Resource 320 provides high-capacity voice over Internet protocol
(VoIP) audio access to the switch for local stations and outside trunks. The IP Media Resource
320 provides audio processing for the following types of calls:
-TDM-to-IP and IP-to-TDM — for example, a call from a 4602 IP telephone to a 6402 DCP
telephone
-IP-to-IP — for example, a non-shuffled conference call
Only one TN2602AP circuit packs are allowed per port network.
Features
The IP Media Resource 320 supports hairpin connections and the shuffling of calls between
TDM connections and IP-to-IP direct connections. The IP Media Resource 320 can also perform
the following functions:
-Echo cancellation
-Silence suppression
-Adaptive jitter buffer (320 ms)
-Dual-tone multifrequency (DTMF) detection
-AEA Version 2 and AES media encryption
-Conferencing
-QOS tagging mechanisms in layer 2 and 3 switching (Diff Serv Code Point [DSCP] and
802.1pQ layer 2 QoS)
-RSVP protocol The TN2602AP IP Media Resource 320 circuit pack supports the following
codecs for voice, conversion between codecs, and fax detection:
- G.711, A-law or Mu-law, 64 kbps
- G.726A-32 kbps
- G.729 A/AB, 8 kbps audio
The TN2602AP also supports transport of the following devices:
-Fax, Teletypewriter device (TTY), and modem calls using pass-through mode
-Fax, V.32 modem, and TTY calls using proprietary relay mode
Note: Note: V.32 modem relay is needed primarily for secure SCIP telephones (formerly known
as Future Narrowband Digital Terminal (FNBDT) telephones) and STE BRI telephones.
- T.38 fax over the Internet, including endpoints connected to non-Avaya systems
- 64-kbps clear channel transport in support of firmware downloads, BRI secure telephones, and
data appliances
The TN2602AP supports STRP media encryption.
Firmware download
The IP Media Resource 320 can serve as an FTP or SFTP server for firmware downloads to
itself. However, this capability is activated by and available for authorized services personnel
only.
I/O adapter
The TN2602AP IP Media Resource 320 circuit pack has a services Ethernet port in the
faceplate. The TN2602AP circuit pack also requires an input/output adapter that provides for one
RS-232 serial port and two 10/100 Mbs Ethernet ports for LAN connections (though only the first
Ethernet port is used).
A family of application-enabling processing platforms that are based on open CPUs and
industry-standard operating systems. Media Servers provide centralized call processing that can
be distributed across a multiprotocol network that includes, but is not limited to, Internet Protocol
(IP). Media Servers support a highly diversified network architecture and provide user
functionality, system management functionality, intelligent call routing, application integration,
mobility, and conferencing.
The Avaya S8730 uses a standard microprocessor engine with a Pentium 4 processor on a
commercial server [8]. S8730 Media Servers support Communication Manager. The S8730-
series media servers use high-speed connections to route voice, data, and video between the
following trunks and lines:
The S8730-series media servers use a Linux platform on an Intel-based server. The S8730-
series media servers are derived from the DEFINITY processor. Nevertheless, the S8730 has
fewer physical components, and provides most of the same features and functionality with
increased capacity.
The total performance of S8730 media server in multi-connect configuration is 300’000 secured
established voice calls in busy hour.
The Centralized reliability and Hardware-duplication configuration is used for Onshore Main
Control Building Telephone Systems. This configuration implies 1+1 redundant (duplicated)
Media server. The Media server duplicated link (over redundant 100Base-T Ethernet and Fibre
optic media) is used for data interchange and synchronization between Media Servers (see Fig.
4-2).
The voice traffic between each PN is carried over IP links using Media Resources boards in
each PN. See also section 4.2.10 TN2602AP IP Media Resource 320 of this document for
further details.
Legend
100Base - T Full
duplex connection
Fibre Optic
Connection
E 1 connection
over SDH
HTTP, Telnet
E 1 connection
over Modem
Craft Terminal
Duplication link
Inter- PH
Inter- PH voice traffic
voice traffic
CNA CNA CNA
Expansion Card
Digital E1 Trunk
Expansion Card
Digital E1 Trunk
Expansion Card
R e s o ur c e
Digital E1 Trunk
R e s o ur c e
R e s o ur c e
M e di a
M e di a
M e di a
IPSI
IPSI
IPSI
Control network1
4x E1
4x E1
2x E1 2xE1
SDH
3x E1
3x E1
2x E1
1x E1
1x E1 HOLD 1
Complex D Complex A OPF Office Construction ITS Camp EWRP Atyrau Power Generation
Station
For more details MCB PABX connections to other PABX(s) via E1 interfaces see 4.4.
The 2 Ethernet switches D-LINK DES-3828 provide connectivity between the servers and the
IPSI circuit pack that reside in each PN. Each of S8730 media servers supports two Ethernet
connections to the Ethernet switches (Figure 4.1), [8].
Each S8730-series media server in a server pair requires a Universal Serial Bus (USB) modem
for maintenance access and to call out an alarm. The modems can share a common phone line
in the designed solution. The online server answers incoming calls. The callers can access the
offline server by means of a telnet session. Each modem connects to a USB port on the media
server. The USB modems used conforms to the Communication Device Class (CDC)
specification, and to the Abstract Control Model (ACM) subclass.
The voice mail service is provided with Avaya Intuity Audix voice mail system. In the INTUITY
AUDIX system, voice messaging is provided by AUDIX software on the system. Subscribers can
record a spoken message, address it, and then send it to other INTUITY AUDIX voice
messaging subscribers. These users can receive the message on their local machine or on
networked INTUITY AUDIX systems.
Additional technical specifications of Voice Mail server can be found in the reference document
[8].
Subscribers instruct the INTUITY AUDIX voice messaging system by pressing the keys on their
touchtone telephones in response to detailed voice prompts from the system.
The Voice Mail System interoperates with Telephone system via analogue telephone lines,
ensuring access to the Voice Mail system voice menu.
The dedicated analogue subscriber lines card is installed in the Media Gateway as per the wiring
diagrams [12].
The incoming analogue telephone lines required to provide voice mail service to the subscribers
are calculated as 0.01 Erlang per subscriber. The below table shows the capacity and the
correspondent amount of voice mail lines required.
-Voice Messaging
-Call Answer
-Automated Attendant
-Bulletin Board
Voice Messaging is similar to an electronic mail system in that messages can be sent to others
without needing to call the recipient directly. The message is then stored in the recipient's voice
mailbox or email mailbox, if applicable. Recipients can access stored messages at their
convenience.
-Automatically place a call from INTUITY AUDIX to the subscriber when there are new
messages waiting.
-Specify the telephone number to be called by INTUITY AUDIX when new messages are
waiting. This telephone number can be for an office, home, or cellular telephone, or for a
pager).
-Record up to nine different personal greetings through the Multiple Personal Greeting feature.
-Play a single greeting for all calls or assign various personal greetings to be played in response
to different types of calls, for example, internal and external, busy and no answer, and/or
out-of-hours.
The Automated Attendant feature can be setup on Voice Mail system but the dedicated
Automated Attendant service is ensured via Voice over LAN (VAL) expansion board ensuring
less loading on Voice mail system and dedicated resources for Automated Attendant service.
See paragraph 4.2.2 Voice Announcement over LAN (VAL) for the details.
The system administrator sets up an automated attendant so that callers hear a menu of options.
Callers then press the button on their telephone keypad that corresponds to the menu option that
they want, and the automated attendant executes the selected option. Those calling from rotary
telephones are typically told that they can hold or call another number to speak with a live
attendant.
An automated attendant menu system, or menu tree, can be designed to contain subordinate
layers of menus or bulletin boards. These submenus, or nested menus, play additional options
that can include a choice that leads to another nested menu.
The voiced menu options that callers hear are actually personal greetings that the subscriber
records for the automated attendant's extension. The text of the message can be changed just
as easily as any personal greeting can. The Multiple Personal Greetings feature can be used to
provide different menus and options for different types of callers.
If the INTUITY AUDIX system has multiple language sets available, the menu options can
include a choice that routes callers to a submenu that is voiced entirely in another language. The
Multiple Personal Greetings feature can also be used to record menus in various languages. For
more information, see Automated Attendants and Bulletin Boards.
A bulletin board is an electronic message system that callers can access to hear messages.
Callers dial the bulletin board telephone number, and the system answers and presents a
recorded message. The major difference between a bulletin board and an automated attendant
is that a bulletin board does not have an option to route to a live attendant. For more information,
see Automated Attendants and Bulletin Boards.
The call logging information is gathered over Data/LAN network infrastructure. Thus the
billing/logging workstation can be setup on special manned locations e.g. control rooms etc.
Avaya Communication Manager organizes and routes voice, data, image and video
transmissions. It can connect to private and public telephone networks, Ethernet LANs, ATM
networks, and the Internet.
Communication Manager is an open, scalable, highly reliable and secure telephony application.
Communication Manager provides user and system management functionality, intelligent, call
routing, self-diagnostics, security, toll fraud protection, and remote access applications.
Avaya Communication Manager feature are delivered over IP. The software can be managed
through the server's existing system administration tools.
(i) Main distribution Frame (MDF) which provides termination of the telephone system
subscriber ports from one side and termination of the backbone copper cables from the
other side. The subscribers’ cables and the backbone cables are typically terminated on
different Krone punch blocks forming thus the permanent non-split connection. The two-
wires twisted pair (included in the scope of supply) is used for interconnection both
sides;
(ii) Digital Distribution Frame (DDF) which provides termination of the telephone system
digital and analogue trunks. The DDF is also used for termination of the digital/analogue
trunks and lines from interface systems
(iii) Intermediate Distribution Frame (IDF) which provides termination of the backbone
cables from one side and Voice Distribution patch panels from another side for remote
locations (which are distant from main Telephone Exchange equipment);
(iv) Voice Exchange/Voice Distribution patch panels which are dedicated for termination IDF
cables and Structure Cabling System (SCS) cables respectively.
The information of cable termination, Krone block allocation and further details can be found in
Termination Schedule, see the reference document [10].
Additional technical specifications of KRONE blocks can be found in the reference document [8]
The Static Transfer Switch (STS) for power sources is used to provide 1+1 power feed
redundancy for the following equipment:
The STS is based on the MGE Pulsar STS 16 model. Pulsar STS handles the automatic or
manual transfer of the critical loads between two independent power sources without interrupting
the supply of power (< 6 milliseconds). Either of the two sources may be designated as the
preferred source with the other becoming the alternate source. In the event of a failure, transfer
from one to the other is automatic and instantaneous. Automatic transfer to the alternate source
takes place if the voltage of the preferred source goes outside a tolerance of 12% above or
below the nominal value. Return to the preferred source is automatic when the voltage returns
within the ±12% tolerance range. To provide a maximum level of protection for the connected
sensitive equipment, both power sources should be on-line type UPSs.
MCB (Main Control Building) and Power Generation Station sites will be connected via these FOM-
4R (first optical multiplexer will be installed at MCB and second optical multiplexer will be installed at Power
Generation Station)
1,2 E1 1,2 E1
PABX PABX
FOM-4R-FE FOM-4R-FE
Figure 4-5 MCB and Power Generator connection via optical multiplexers
Features
Manufacturer NGNSystems (Canada)
Model FOM-4R, V2
E1 line Interface
4 E1 links
Line Rate 2.048 Mbps
E1 Impedance 120 Ω (balanced). Connector RJ-45
Physical/Electrical
Height 44 mm (1U), Width 485 mm, Depth 160 mm
Mounting Stand-alone, 19” inch rack mount
Weight 1,2 кг
Power Source 220 VAC (50H/60Hz)
Power Consumption 10-40 W
CLAN 1. 1
2 xS 8730
MED RES1. 1
G 650
MED RES2. 1
MED RES3. 1
VAL
CLAN2.1
IPSI A S 8730 A
S 8730A S 8730 B
Voice mail
VoIP Traffic
VLAN00.2
192. 168. 20.0/ 24
Billing Server
Call Processing A System Access
( CNA ) KZOP- PBX -82 VLAN
VLAN 01.1 10. 191. 50. 64/26
198.152. 254.0 / 24
Administration
Workstation
Control Network A
2xEthernet
. Switches
The Project SDH system will be used as a transmission media for PABX System voice
communications between the locations. The SDH system will provide clear point-to-point E1
connections on each of the locations. The SDH system E1 channels are to be configured as per
the following diagram.
6 . Complex D 1 . CCB
n 18
3xE1 8 . EWRP PSTN
2x
E1
3xE1 n20
E1
1x
1xE1
1xE1 2 . Atyrau
2xE1
n24
5 . Complex A 3xE1
4 . OPF
Construction 3xE1
Office
7 . Power 3xE1
Generation
Station
3 . ITS Camp
E1 over SDH
E1 over Modems
Note CCB and Power Generation Station sites are connected via media modems (optical
multiplexers). Media modems provide NON-SDH connection from CCB to Power Generation
Station PABX.
Note E1 connection between CCB and Caspian Yard is not included in the SOW
Note Future E1 connection between CCB and EWRP will be provided by other
Note nxy is the undefined full capacity between location X and Y (no information available). The
specified in blue is the E1 capacity between the locations. The analogue trunks are not used for
inter-PABX links.
Note that the E1 channels are calculated only for the Telephone Exchanges within Contractor’s
scope of supply.
The amount of voice channels required for each Telephone System is calculated as 0.25 Erlang
per subscriber (one voice channel per four subscribers per one hour). The following table
represents the voice channel calculation and the number of E1 bearers (one E1 bearer carries
30 duplex voice channels).
The analogue trunks capacity is specified in the Reference Document [7], the analogue trunks
shall be used for additional (radio etc.) systems connection. The digital (E1) trunks are used for
inter-PABX connection.
Max. Power
Node Fuse
consumption Note
rate
Tag number Designation Feed A Feed B
A1-9900-NTX-020-001A Media Server A 600W 4A
A1-9900-NTX-020-001B Media Server B 600W 4A
A1-9900-NTS-020-001 A Ethernet Switch 60W 60W 2A Via STS
A1-9900-NTS-020-001 B Ethernet Switch 60W 60W 2A Via STS
A1-9900-NTZ-020-010 Voice Mail Server 300W 300W 4A Via STS
- Rack Fan Unit 30W 30W 2A Via STS
A1-9900-NTU-020-001 Static Transfer Switch - - 16A
Total Max. loading 1050W 1050W
Note: the equipment powered via STS loads only one feed in the same instant.
Max. Power
Node Fuse
consumption Note
rate
Tag number Designation Feed A Feed B
Duplicated
A1-9900-NTM-020-001 Media Gateway 01 600W 600W 6A
power supply
Duplicated
A1-9900-NTM-020-002 Media Gateway 02 600W 600W 6A
power supply
Duplicated
A1-9900-NTM-020-003 Media Gateway 03 600W 600W 6A
power supply
Duplicated
A1-9900-NTM-020-004 Media Gateway 04 600W 600W 6A
power supply
- Rack Fan Unit 30W 2A
Total Max. loading 2400W 2430W
Note: the Media Gateways uses duplicated power supplies and load only one feed in the same
instant. The switchover from one power supply to another is done automatically upon active
power supply failure. The total max loading specified for each feed is for fully loaded feed. The
actual loading shall be equal to the sum of the active power supplies power consumption.
Max. Power
Node Fuse
consumption Note
rate
Tag number Designation Feed A Feed B
Duplicated
A1-9900-NTM-020-005 Media Gateway 05 600W 600W 6A
power supply
Duplicated
A1-9900-NTM-020-006 Media Gateway 06 600W 600W 6A
power supply
- Rack Fan Unit 30W 2A
Total Max. loading 1230W 1200W
Note: the Media Gateways uses duplicated power supplies and load only one feed in the same
instant. The switchover from one power supply to another is done automatically upon active
power supply failure. The total max loading specified for each feed is for fully loaded feed.
Note the rack Lamps are powered from non-UPS feed as they are non-critical equipment. The
rack lamps are daisy-chained (in parallel way so one of the lamp’s failures shall not affect
others). Each lamp requires 14W when switched on. The total max power for the Rack A1-9900-
NTR-020-001…004 is 64W. The rack lamp is switched on automatically upon rack door opening.
Note: the rack fan units in the racks A1-9900-NTR-020-002 and A1-9900-NTR-020-003 are
powered from different feeds and ensure hot air exhaust upon any of the feed’s failure. This is
provided also with no side panel installed between the mentioned racks.