वर्गीय आव�यकताएँ सं.:टीईसी/जीआर/एसडब쥍यू/पीबीए啍स-001/02/अ啍टूबर-17 (सं.:टीईसी/जीआर/एसडब쥍यू/पीबीए啍स-सीपीए/01/फरवरी-09 को अधिक्रधित करता है) GENERIC REQUIREMENTS
No. : TEC/GR/SW/PBX-001/02/OCT-17 (Supersedes No. TEC/GR/SW/PBX-CPA/01/FEB-09)
प्राइवेट ऑटोिैटटक 녍ांच ए啍सचᴂज
PRIVATE AUTOMATIC BRANCH EXCHANGE
© टीईसी, 2017 © TEC, 2017
इस सवााधिकार सुरधित प्रकार्शन का कोई िी धहसा, दूरसंचार अधियांधिकी कᴂद्र, नई दद쥍ली की धलधखत ,र्गﴂवीकृधत के धबना, दकसी िी 셂प िᴂ या दकसी िी प्रकार से जैसे -इले啍रॉधनक, िैकेधनकल,फोटोकॉपी, टरकॉ셍ड कैननंर्ग आदद 셂प िᴂ प्रेधित, संग्रहीत या पुन셁配पाददत न दकया जाए । All rights reserved and no part of this publication may be reproduced, stored in a retrieval
system or transmitted, in any form and by any means - electronic, mechanical, photocopying, recording, scanning or otherwise, without written permission from the Telecommunication Engineering Centre, New Delhi.
दूरसंचार अधियांधिकी कᴂद्र खुर्शीदलाल िवन, जनपथ, नई दद쥍ली–110001, िारत TELECOMMUNICATION ENGINEERING CENTRE KHURSHIDLAL BHAWAN, JANPATH, NEW DELHI–110001, INDIA www.tec.gov.in
Release 04: OCT, 2017 Price: 800/-
FOREWORD
GR No. TEC/GR/SW/PBX-001/02/OCT-17 2
Telecommunication Engineering Centre(TEC) functions under Department of Telecommunications (DOT), Government of India. Its activities include: Issue of Generic Requirements (GR), Interface Requirements (IR), Service Requirements (SR) and Standards for Telecom Products and Services Field evaluation of products and Systems National Fundamental Plans Support to DOT on technology issues Testing & Certification of Telecom products For the purpose of testing, four Regional Telecom Engineering Centers (RTECs) have been established which are located at New Delhi, Bangalore, Mumbai, and Kolkata.
ABSTRACT
This document covers the Generic Requirements (GR) of PABX connected to Public Switched Telephone Networks (PSTN) or IP core (soft-switch/CSCF) of Indian Telecom Network. This document covers the Generic Requirements of PABX for supporting speech communication, voice-band applications, data communication, video communication and multi-media applications. It covers all types of interfaces for connection to public network as well as essential PABX facilities, features and its performance requirements.
CONTENTS
GR No. TEC/GR/SW/PBX-001/02/OCT-17 3
Clause Particulars Page No.
History Sheet …………………………….………………………………...... …… 5 References…………………………………………………………………….…… 6
Chapter 1 1.0 Introduction ……………………..……….…………...... 9
2.0 Description…………………………………....………………..…….…11
3.0 Functional/Operational Requirements .……………...... 13
4.0 Interface Requirements……………………….…………………..….. 35
5.0 Quality Requirements ………………………………………….….…. 45
6.0 EMI/EMC Requirements ……………….……………………………46
7.0 Safety Requirements………………………………………………. … 51
8.0 Security Requirements……………………………………………….. 52
Chapter 2
9.0 Information for the procurer of product……………...... 53
10.0 Specific items to be mentioned in the certificate ...... 97
Abbreviations ……………………………………………………..…. ..98
GR No. TEC/GR/SW/PBX-001/02/OCT-17 4
HISTORY SHEET
S. GR No. Title Remarks No. 1. TEC/GR/SW/PBX GR of PABX Issue 01 -CPA/01/FEB-09 It covers all types of user terminals namely Analogue, ISDN, Wireless, IP etc. Covers all types of interfaces for connection to public network including IP interface. Covers Mandatory as well as desirable requirements. 2. TEC/GR/SW/PBX PABX Issue 2 -001/02/JUL-17 Prepared in new GR format __MDF related clauses are removed __ Traffic related clause are removed __ __socket address added in CDR __Toll by pass issue added __IP related RFC added, ___ In place of TEC IR No. TEC/IR/NNI- IPN/01 only relevant clauses are added __ Rephrase and added some of the clauses to bring more clarity
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REFERENCES
S.No. Document No. Title/Document Name (I) : TEC GR/IR/SDs 1. TEC/SD/DD/EMC- Electromagnetic Compatibility Standard for 221/05/OCT-16 Telecommunication Equipment 2. SD/ISN-01 National standard for ISDN Basic rate access 3. SD/ISN-02 National standard for ISDN Basic rate access 4. TEC/SD/SW/VAN- National Standards for V5.2 Interface SIG 5. TEC/IR/MFR- Multi Frequency Register Signaling (R2MF) SIG/01/JAN-09 (II) : ITU / ETSI/ IEEE Standard/Recommendations 1. ITU-T G.703 Physical/electrical characteristics of hierarchical digital interfaces 2. ITU-T G.704 Synchronous frame structures used at 1544, 6312, 2048, 8448 and 44 736 kbit/s hierarchical levels 3. ITU-T G.711 Pulse code modulation (PCM) of voice frequencies 4. ITU-T G.723 Speech codec using Adaptive Differential Pulse Code Modulation (ADPCM) 5. ITU-T G.729 Coding of speech at 8 kbit/s using conjugate- structure algebraic-code-excited linear prediction (CS-ACELP) 6. ITU-T G.823 The control of jitter and wander within digital networks which are based on the 2048 kbit/s hierarchy
7. ITU-T T.38 Procedures for real-time Group 3 facsimile communication over IP networks 8. ITU-T H.323 Packet-based multimedia communications
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systems 9. Q.23 Technical features of push-button telephone sets 10. Q.24 Multi-frequency push-button signal reception 11. Q.552 Transmission characteristics at 2-wire analogue interfaces of digital exchange 12. Q.920 ISDN user-network interface data link layer - General aspects 13. Q.921 ISDN user-network interface - Data link layer specification 14. Q.931 ISDN user-network interface layer 3 specification for basic call control (III) IETF Recommendations 1. RFC 791 IPv4 2. RFC 2460 IPv6 3. RFC 2119 Key words for use in RFCs to Indicate Requirement Levels (IV) : Other Standards 1. CISPR 11 Limits and methods of measurement of radio disturbance characteristics of industrial, scientific & medical (ISM) radiofrequency equipment 2. CISPR 22 Limits and methods of measurement of radio disturbance characteristics of ITE 3. IEC/EN 61000-4-2 Testing and measurement techniques – Electrostatic discharge immunity test 4. IEC/EN 61000-4-3 Testing and measurement techniques – Radiated, radio-frequency, electromagnetic field immunity test 5. IEC/EN 61000-4-4 Electromagnetic compatibility (EMC) - Part 4-4: Testing and measurement techniques - Electrical fast transient/burst immunity test 6. IEC/EN 61000-4-5 Electromagnetic compatibility (EMC) - Part 4-5: Testing and measurement techniques - Surge
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immunity test 7. IEC/EN 61000-4-6 Electromagnetic compatibility (EMC) - Part 4-6: Testing and measurement techniques - Immunity to conducted disturbances, induced by radio- frequency fields 8. IEC/EN 61000-4-11 Electromagnetic compatibility (EMC) Part 4-11: Testing and measurement techniques Voltage dips, short interruptions and voltage variations immunity tests 9. IS 13252 / IEC Information Technology Equipment -- Safety, Part 60950 1: General Requirements 10. EN 55011 Industrial, scientific and medical (ISM) radio- frequency equipment - Electromagnetic disturbance characteristics - Limits and methods of measurement 11. EN 55022 Information Technology Equipment - Radio disturbance characteristics - Limits and methods of measurement
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CHAPTER-1 1.0 Introduction 1.1 Private Automatic Branch exchange (PABX)is an in-house telephone switching system which makes connections among the internal telephones of a private organization and also connects them to the public telecom network via various interface(s) as depicted in Fig-1 of this document. PABX can have various types of extensions such as analogue phone, digital Phone, IP Phone, etc.
1.2 This document covers the Generic Requirements (GR) of PABX connected to Public Switched Telephone Networks (PSTN) or IP core (soft- switch/CSCF) of Indian Telecom Network. This document covers the Generic Requirements of PABX for supporting speech communication, voice-band applications, data communication, video communication and multi-media applications. It covers all types of interfaces for connection to public network as well as essential PABX facilities, features and its performance requirements. 1.3 Approval of equipment against this GR shall not be construed as an authorization to evade surreptitiously, regulations including toll-bypass concerning the use of telecom services. Functioning or intended use of the equipment shall conform to the prevailing laws/regulation/instructions of Govt. of India. 1.4 Approval of IP interface against this GR, does not entitle the user to connect the equipment to the network of Internet Service Provider’s (ISP). 1.5 This GR of PABX is used where cards are inserted in the motherboard slot depending upon the user requirement 1.6 (a) For all ITU–T/ETSI/3GPP/IEEE recommendations, TEC standards/ specification and other standards referred in this document, the latest release/issue with all associated amendments, addendum and corrigendum shall be applicable. (b) The RFC documents of IETF are subject to periodic revision. Hence, where ever RFCs are mentioned in this document, the offered product shall
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meet either the referred RFC or its previous version or its previous draft or its updated version. Wherever a feature of RFC is mentioned, the product shall comply with the part of RFC specifying the feature. (C) For all IETF RFCs, the interpretation of clauses of RFCs shall be as per RFC 2119.
1.7 All the requirements described in chapter 2 of this document are suggestive requirements and shall be decided by the purchaser at the time of procurement/ tender as per his requirements. However, the requirements described in Chapter-2 will not be tested/ verified by TEC.
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2.0 Description
2.1 This GR is intended to facilitate the verification of the capability of the equipment for correct operation and satisfactory inter working with the exchanges presently working in the Indian Telecom networks. The PABX shall be compatible with various technology exchanges presently working in the Indian Telecom networks for satisfactory working. 2.2 Various types of extensions and network interfaces used in PABX are shown in Fig.1. 2.3 Interfaces towards Public Network PABX shall be connected to TDM or IP based Public Telephone Network of Indian Telecom Network through Interfaces described in this clause. The applicant may seek type approval for one or any combination of various types of interfaces given below in Table 1: Table 1. Interfaces toward Public Network
S.No. Interface Details
2W DEL: 2-wire DELs for outgoing calls as well incoming (i) TYPE-1 calls. The junctions of PABX will be terminated on the analog subscriber line cards of the main exchange. E1R2 Interface i.e. 2048 kbit/s using Indian R2 signalling as (ii) TYPE-2 per TEC IR No. TEC/IR/MFR-SIG ISDN Basic Rate Access at S-Interface BRI as per National (iii) TYPE-3 standard for ISDN Basic rate access No. SD/ISN-01 and shall terminate on the ISDN BRI access card. ISDN Basic Rate Access at U-Interface BRI as per National (iv) TYPE-4 standard for ISDN Basic rate access No. SD/ISN-02 and shall terminate on the ISDN BRI access card. ISDN Primary rate access interface (PRI i.e. 30 B + D) as per (v) TYPE-5 TEC standard No. SD/ISN-01 and shall terminate on the ISDN PRI access card. V5.2 Interface as per National Standards TEC/SD/SW/VAN- (vi) TYPE-6 SIG
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(vii) TYPE-7 Ethernet based Interface as per clause 3.18 of this document.
Wire-less (DECT)
2W DEL
E1R2 BRI S- TDM Base station PABX interface NETWORK BRI U-
interface PRI POTS V5.2
ISDN NT
S Softswitch
Ethernet IP PHONE Core IP MM PC Network
Fig. 1. Scenario for PABX deployment
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SECTION-A (For all Interfaces)
3.0 Functional & Operational Requirement
3.1 General All the requirements described in this section, shall be complied by all PABX irrespective of types of interfaces and/or the type of extensions for which approval is sought. Additional requirements for different types of interfaces are given in subsequent sections. 3.1.1 The applicant shall submit the list of all units which are part of PABX for which the Type approval is sought. The equipment shall be complete in itself along with associated power supply arrangement 3.2 Functionalities 3.2.1 The equipment shall provide interoperability with all types of exchanges in the present Indian Telecom Network.
3.2.2 The equipment shall not necessitate any changes in the existing Public Switched Telephone Network (PSTN)/IP network and shall not impact the interfaces and protocols in these networks.
3.2.3 The calls within PABX system shall be switched without using link with Indian Telecom Network. Intra-PABX calls shall not be affected in case of link failure to Indian Telecom Network.
3.2.4 When the links between Public network and PABX are out of service, PABX shall send suitable announcement/tone/message to calling extension subscriber. Intra-PABX calls shall not be affected in case of link failure to Indian Telecom Network.
3.2.5 Whenever any extension goes off-hook, the PABX shall feed dial tone to its extensions (except in case of IP extension).
3.2.6 It shall be possible to make calls to emergency numbers (e.g. 100, 101, 102 112 etc) and IN calls from all types of user terminals i.e. extensions provided. PABX shall support transmission of DTMF as per ITU-T Recommendation Q.23 {In case of interface(s) type (1) and type (2)} and as per IEEE Recommendation SIP info or RFC2833/4733 {in case of
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Type vii interface} in the post dial scenario” . For the rest of the interfaces a information /signal/protocol correspond to DTMF signalling shall be transmitted over the network. 3.2.7 The PABX shall not provide for any feature/ facility and interface that could adversely affect the working of the Indian Telecom Network.
3.2.8 It shall be possible to store at least 24 digits (excluding the ‘0’ dialled for Public network access), dialed by any extension number and PABX shall be able to send/ forward these digits to the main exchange, whenever required. It shall be possible to forward, up to 24 digits (as per requirement), for all types of interfaces.
3.2.9 The PABX shall support calling the PSTN telephones by their E.164 address.
3.2.10 The PABX shall provide for the facility of the extension user to transfer the incoming call from the main exchange to other extensions directly (i.e. without the assistance of PABX attendant/operator).
3.2.11 The PABX shall provide for time out and release of call under held up condition during pre-dialling, dialling, post-dialling, ringing phase and Called Subscriber Held (CSH) condition.
3.2.12 For all types of calls to Public network, Calling Line Identification (CLI) shall be forwarded to the main exchange including calling extension number (except for Type-1 interfaces towards public network). In case of call from public network, CLI received from main exchange shall be forwarded to called extension number.
3.2.13 It shall not be possible to modify the CLI in PABX. In addition, the applicant/ manufacturer/licensee shall give undertaking that there is no software available in their system for providing such function/service.
3.2.14 Terminating Free Number: It shall not be possible to make call from public network to any extension as charge free call.
3.2.15 It shall be possible to connect any TEC approved phones/terminals of any make without affecting the system functioning
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3.3 Alarm/Fault Diagnosis of Interface
The PABX shall provide facility for isolating the faults on the PABX side or exchange side for any particular trunk/ link and/or all trunks/ links. i In case of non-allotment of any trunk to an extension due to any faulty condition detected by the PABX, a suitable alarm/indication for the particular trunk shall be provided.
ii It is desirable that the PABX should provide for indication of the status of each trunk on the operator's console or a suitable visual display on the PABX system.
iii It shall be possible to run diagnosis with man-machine command to test the junctions to/from main exchange.
3.4 Maintenance/Operator Console:
PABX should support maintenance/ configuration of the complete system. It shall be possible to change the configuration of the system and do troubleshooting. The system errors/alarms like PRI is out-of-service, IP trunk not getting registered, etc. shall be logged in the system fault log. The system fault log can be stored in the system or can be sent to any of the networked computer in the customer’s premise for backup. 3.5 Call Detail Records:
PABXs shall provide Call Detail Record (CDR) for all calls from extensions to Public network and from Public network to PABX extensions. Facility shall exist to take a print out for each individual extension. At least the following information shall be given in a CDR:
(i) Type of call (originating or terminating)
(ii) Date and time of answer (in case of successful call)
(iii) Calling subscriber identity.
(iv) Socket address i.e. Port no. & IP address of calling subscriber in case call is made from IP terminal.
(v) Incoming/Outgoing junction number
(vi) Number dialled by the calling subscriber i.e. called number
(vii) Duration of conversation in multiple of seconds.
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(viii) Indication of Type of supplementary services used during the call e.g. normal call, conference call, call forwarding (for incoming call only).
(ix) Redirecting number in case of call forwarding for incoming call only. 3.5.1 Provision shall exist for storing the CDR data on the system and to transfer the same on any other external device/system. Applicant shall declare the CDR storing capacity of the system. 3.5.2 Applicant shall supply the required information in the human readable format and the ‘data structure’ to monitoring agencies. Required software may be part of the PABX system or may be supplied as an external entity. 3.5.3 Neither it shall be possible to edit the CDR file nor it shall be possible to delete CDR file from the system.
3.6 Command log Commands which are used for modification of PABX program or data shall be stored in the system. Provision shall exist for storing the command log/history log on the system or any other external device/system. It shall not be possible to modify or delete log file. Applicant shall declare the storing capacity of log file in the system. 3.7 Documentation The manufacturer / supplier shall supply at least one copy of the equipment literature in English. It shall include system description, user manual and complete specifications of the equipment in digital format. 3.8 IP PABX (Applicable only for IP-PABX) In case PABX is based on IP technology, in addition to various functionalities described in this document, following functionalities shall also be complied:
3.8.1 PABX shall support interworking with PSTN (applicable only if applicant seeks approval for S-interface/U-interface/PRI-Interface/2W-Interface/MFR2- interface/V5.2 interface.)”
3.8.2 Bearer Independent Call Control (BICC) and SIP inter working support in MGC shall be based on the ITU-T recommendation Q.1912.5 (Applicable only if the applicant seeks approval for S-interface/U-interface/PRI interface)
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3.8.3 Connections to the PSTN shall be made through SIP based PSTN gateway controller (e.g. MGC) that translates SIP messages into PSTN message formats. (Applicable only if the applicant seeks approval for 2W-interface/MF R2-interface/V5.2 interface)
3.8.4 System shall provide encryption and encapsulation method for data security. (Applicable only if the applicant seeks approval for Ethernet interface) 3.8.5 Ethernet connections used shall be in conformity to ITU-T recommendations IEEE 802.3 (Applicable only if the applicant seeks approval for Ethernet interface) 3.8.6 Addressing: The PABX shall support IP addressing as per version 4 (IPv4) as well as version 6 (IPv6). (Applicable only in case of approval for IP connectivity) 3.8.7 PABX shall have the capability to inter-work with IP networks supporting IPv4 as well IPv6. (Applicable only in case of approval for IP connectivity)
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SECTION -B Functional Requirements (For Type-1 Interface only)
3.9 Applicable for Type-1 Interface only In addition to functional requirements described in section A of this chapter, this section specifies the requirements for PABXs with TYPE-1 interface (2- wire analog subscriber access interfaces i.e. DELs) both for outgoing calls from PABX to PSTN and incoming calls from PSTN. 3.9.1 General: The PABX shall have the capability to interconnect with the PSTN Exchange (also referred as main exchange in this document) using 2-wire analog subscriber access interfaces (DELs) for originating as well as terminating calls, with the provision for reversal and periodic metering pulses of 16 KHz., The junctions of PABX (i.e. DELs) will be terminated on the analog subscriber line cards of the PSTN exchange. 3.9.2 Whenever any extension goes off-hook, the extension shall get dial tone. 3.9.3 The PABX shall not leave the 2W DEL unconnected under any state other than total system failure, as this is an unhealthy condition for the main exchange leading to alarm conditions, repeated dialing, loss of calls etc. 3.9.4 Access to Public Network: The Public Network shall be accessed by dialing single digit. Dialling of `0' (or any other suitable code) shall be used to provide access to the Public network (O/G junctions) and the junction access should have a random or rotating priority to distribute the traffic. Normally, after receipt of digit ‘0’, PABX shall put through the call to main exchange which shall feed second distinctive dial tone to calling extension subscriber. Further digits shall be dialled by the caller after receipt of second dial tone. 3.9.5 Line Hunting For outgoing calls, the PABX shall provide access to the public network using sequential or random search for selection of trunk/link towards Indian Telecom Network.
3.9.6 Direct Inward Dialing (DID ) Facility The PABX shall support the facility where any subscriber from public network can dial the PABX extension number. It can be implemented in either of the following methods: (a) Without the assistance of any PABX operator. In such scenario, on
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receipt of incoming call, PABX shall feed second dial tone/ announcement after which the calling subscriber shall dial extension number. (b) With the assistance of Operator- In this case, any incoming call shall be connected to operator position and the operator shall be able to transfer the call to desired extension. 3.10 Switching: (a) The connection from PABX to the main exchange shall be independent of line polarity. (b) The PABX shall be capable of detecting the calling condition (I/C call) by sensing the ringing signal (75 ± 5 V rms, 25 Hz) from main exchange. (c) The PABX shall be capable of working satisfactorily on the lines from exchange under adverse conditions of leakage i.e. insulation resistance as low as 20 k between 'a' and 'b' limbs and to earth. (d) It shall be possible to conduct the following tests from the main exchange. PABX shall provide appropriate termination on its outgoing junctions, for conducting the following line testing from main exchange: (i) Continuity or click test as in the case of a normal telephone having capacitor 2 F in series with the bell. (ii) Insulation resistance test of 'a' and 'b' limbs of the line with respect to earth and between each other. The insulation resistance shall be more than 250 k during the idle conditions of the PABX junctions. These tests shall be possible under all conditions of the PABX such as power down, night switching etc. (e) The PABX shall provide for local battery feed to the extensions. The equipment shall also provide for DC isolation between the DC supplies used in the system and the DC battery feed from the exchange i.e. exchange battery shall not be used for PABX circuitry. Either the +ve or -ve end of the local battery feed to the extensions should be connected to the general earth. (f) In the case of outgoing calls from PABX towards the exchange, disconnection of call from the PABX side shall be transmitted towards the exchange as an ‘open line condition’ for a minimum period of 1 sec.
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(g) PABX shall ensure release of the established connection before allotting the junction to another extension. (h) At the end of an incoming call, the minimum period of busying a junction against access by another extension shall be 2 seconds. (i) Incoming junction calls to PABX from main exchange that have been extended through the PABX operator’s console to any extension shall either be returned back to the attendant/operator (In case of a PABX not having an operator, an extension that has originally received the call shall be assumed to be the operator) or a suitable tone/announcement shall be returned to calling subscriber, if the called extension does not answer within a pre-determined time. (j) It shall be possible to provide the facility of one way working of the junctions i.e. some of the junctions from the main exchange to be used as I/C junctions only and should not be accessed for any O/G call from the PABX to the main exchange. (k) PABX Shall ensures a minimum current of 22mA on junction/trunk line. In any case, current drawn by PABX shall not exceed 60 mA on junction/trunk line. 3.11 Transmission Parameter 3.11.1 The PABX shall conform to the following transmission parameters for 2Wire interface of PABX available at the MDF / Terminations or equivalent distribution point. The specifications for all the transmission parameters are given below. The nominal value of the impedance at the line access is 600 , balanced.
3.11.2 Return Loss: The return loss measured across the nominal impedance of 600 should not be less than the following values: Frequency Return loss 300 Hz 14 dB 400 Hz 16 dB 500 Hz 18 dB
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500 Hz-2000Hz 18 dB 2700Hz 16 dB 3400Hz 14 dB
3.11.3 Impedance unbalance about earth: The value of longitudinal conversion loss (LCL) should exceed the minimum values given below: Frequency Longitudinal conversion loss (LCL) 300-600 Hz 40 dB 600-3400Hz 46 dB 3.11.4 Cross Talk The cross talk between individual lines should be such that with a sine wave signal of frequency at a level of 0 dBmo applied to a port, the cross talk level received in any other line should not exceed –65 dBmo.
3.11.5 Idle State Noise Single Frequency Noise- The level of any single frequency (in particular the sampling frequency and its multiples) measured selectively should not exceed –50 dBmo.
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SECTION – C (For Type-2 Interface only) 3.12 E1 R2 Interface In addition to functional requirements described in section A, this section specifies the requirements for PABXs with TYPE-2 interface (E1 R2) both for outgoing calls from PABX to Public network and incoming calls from the Public network using Indian R2 signalling. 3.12.1 General: The PABX shall have the capability to interconnect with network using E1 i.e. 2048 kbit/s PCM junctions. 3.12.2 The O/G and I/C Junctions of the PABX shall be terminated on the inter exchange trunk interface of main exchange. 3.12.3 Whenever any extension goes off-hook, the PABX shall feed dial tone to its extensions.
3.12.4 Depending on the capacity of the PABX, the exchange will send all/last few digits of called number (as per local requirements) to the PABX. 3.12.5 Access to Public Network: The Public Network shall be accessed by dialing single digit. Dialling of `0' (or any other suitable code) shall be used to provide access to the Public network (O/G junctions) and the junction access should have a random or rotating priority to distribute the traffic. Normally, after receipt of digit ‘0’, PABX shall put through the call to main exchange which shall feed second distinctive dial tone to calling extension subscriber. Further digits shall be dialled by the caller after receipt of second dial tone. 3.12.6 The PABX shall provide for time out and release of outgoing junction/circuit under ‘held up condition’ during pre-dialling, dialling, post-dialling, ringing phase and CSH condition. 3.13 Calling Line Identification 3.13.1 For all types of calls from Public Network, Calling Line Identification (CLI) of a pre-defined number of digits will be sent by the main exchange. At present it is 10 digits CLI, but it can be increased up to 16 digits. The PABX shall be able to receive complete Calling Line Identification (CLI), (consisting of country code + Area code + directory number) from the main exchange (at present country code is not included in CLI). It shall not be
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possible to change the CLI in PABX. 3.13.2 For all types of calls made from extension to public network (If extension numbering scheme used is from ‘group of directory numbers’ allotted by service provider), CLI of extension number shall be send to the main exchange. 3.13.3 Direct Inward Dialling (DID) Facility The PABX shall support the facility where any Public network subscriber shall be able to dial the PABX extension number without the assistance of any PABX operator/attendant. This facility can be implemented by any one of the following methods: i. Direct Inward Dialling (DID): Service provider shall allot ‘a group of directory numbers’ from the concerned parent exchange to the PABX. All extension users will have directory numbers as per E.164 numbering plan of the service provider. In such cases, main exchange shall send digits of extension number to PABX. ii. Each PABX shall be allotted a single directory number by the service provider. When a Public network subscriber dials PABX number, PABX shall feed announcement to caller and calling subscriber shall dial the extension number 3.13.4 Clock: The PABX shall extract the clock from 2048 kbit/s link between the main exchange and the PABX and shall use it for timing purpose. 3.13.5 Transmission Parameters (as per ITU-T Recommendations G.703, G.704 and G.823)
i). Nominal bit rate: 2048 kbit/s. ii). Bit rate accuracy: ±50 ppm (±102.4 bit/s). iii). Code: High density bipolar of order 3 (HDB3) iv). Specifications at the output ports
Nominal peak voltage of a mark (pulse) 3 V±0.3 V Peak voltage of a space (no pulse) 0±0.3 V Nominal pulse width 244 ns
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M Ratio of the amplitudes of positive and 0.95 to 1.05 a negative pulses at the centre of the pulse
x interval Ratio of the widths of positive and negative 0.95 to 1.05 i pulses at the nominal half amplitude m
um peak-to-peak jitters at an output port
Measurement bandwidth, Peak-to-peak Jitter –3 dB frequencies (Hz) amplitude (UI) ** 20 to 100 k 1.5 18 k to 100 k 0.2 **1 UI = 488 ns
v). Transmission loss at the input ports shall be in the range 0 to 6 dB at a frequency of 1024 kHz.
vi). Jitter tolerance at input port should be as indicated below
Frequency Jitter amplitude to be applied Peak to Peak ( UI ) ** 20 Hz 1.5 2.4 KHz 1.5 18 KHz 0.2 100 KHz 0.2 ** 1 UI (Unit Interval) = 488 ns. vii). The return loss at the input port should have the following minimum values:
Frequency range (kHz) Return loss (dB) 51 to 102 12 102 to 2048 18 2048 to 3072 14
3.13.6 Transmission Parameters (as per ITU-T Recommendation Q.552, applicable only if approval for analogue extensions is also sought)
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i) Transmission Loss The transmission loss provided by the PABX for the input half connection (A-D) shall be in the range of 0 to 2.0 dB and for output half connection (D-A) in the range of 0 to 8 dB. ii) Weighted Noise The total weighted noise level for the input half connection (A-D) shall not exceed -63.5 dBmop to -64.2 dBmop for input relative level of 0 dBr to 2.0 dBr respectively and the total weighted noise level for the output half connection (D-A) shall not exceed -66.4 dBmop to -66.9 dBmop for output relative level of 0 dBr to -8.0 dBr respectively. iii) Total Distortion Total distortion including quantization distortion for input and output half connection shall be better than the following values:
Input level Signal to total distortion ratio for input half connection output half connection 0 dBmO 35.0 dB 35.0 dB -10 dBmO 35.0 dB 35.0 dB -20 dBmO 35.0 dB 33.8 dB -30 dBmO 32.9 dB 28.8 dB -40 dBmO 24.9 dB 19.5 dB -45 dBmO 19.9 dB 0.5 dB
iv) Group Delay Group delay between Digital to Analog interfaces shall meet the following requirements: Frequency Permitted limit (Hz) (micro second)
500 1950
604 1950
1000 1950
1792 1950
2604 1950
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2792 1950
Note: 1950 micro seconds is the mean value. However, the limit for 0.95 probability of not exceeding is 2700 micro seconds.
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SECTION – D (For Type-3 Interface only) 3.14 BRI at S-Interface In addition to functional requirements described in section A, this section specifies the interface requirements for PABX with Type-3 interface (ISDN BRI at S interface) both for outgoing calls from PABX to PSTN/ISDN Network and incoming calls from the PSTN/ISDN Network. 3.14.1 The PABX shall have the capability to interconnect with PSTN/ISDN using ISDN Basic Rate Access (2B+D) at S-Interface. This interface is terminated on S-interface port of PABX. 3.14.2 Whenever any extension goes off-hook, the PABX shall feed dial tone to its extensions. 3.14.3 The PABX shall not leave the I/C trunks un-connected under any state other than total system failure, as this is an unhealthy condition for the main exchange leading to alarm conditions, repeated dialling, loss of calls etc. 3.14.4 Access to Public Network: The Public Network shall be accessed by dialing single digit. Dialling of `0' shall be used to provide access to the Public Network (O/G junctions) and the junction access should have a random or rotating priority to distribute the traffic. Normally, after receipt of digit ‘0’, PABX shall put through the call to main exchange which shall feed second distinctive dial tone to calling extension subscriber. Further digits shall be dialled by the caller after receipt of second dial tone. 3.14.5 Direct Inward Dialling (DID) Facility The PABX shall support the facility where any subscriber from public network can dial the PABX extension number. It can be implemented in either of the following methods: (a) Without the assistance of any PABX operator. In such scenario, on receipt of incoming call, PABX shall feed second dial tone/ announcement after which the calling subscriber shall dial extension number. (b) With the assistance of Operator- In this case, any incoming call shall be connected to operator position and the operator shall be able to transfer the call to desired extension.
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3.14.6 Depending on the requirements of the PABX, the exchange will send all/last few digits of called number to the PABX. The PABX shall indicate the end of digits by sending appropriate signal to the main exchange.
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SECTION- E
(For Type-4 Interface only)
3.15 BRI at U-Interface In addition to functional requirements described in section A, this section specifies the interface requirements for PABX with Type-4 interface (ISDN BRI at U-interface) both for outgoing calls from PABX to Public network and incoming calls from the Public network. 3.15.1 The PABX shall have the capability to interconnect with PSTN/ISDN using ISDN Basic Rate Access at U-Interface. This interface is terminated on U- interface port of PABX. 3.15.2 Alarm/Fault Diagnosis of Interface: The PABX shall provide suitable test functions like loop back for test commands given by exchange in case of BRI U-Interface. 3.15.3 Clause no 3.14.2 to 3.14.6 mentioned in Section-D (BRI S-interface) shall also be applicable for BRI U-interface.
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SECTION –F (For Type-5 Interface only)
3.16 PRI Interface In addition to functional requirements described in section A, this section specifies the functional requirements for PABX with Type-5 interface (ISDN PRI i.e. 30B+D) both for outgoing calls from PABX to Public network and incoming calls from Public network. 3.16.1 The PABX shall have the capability to interconnect with PSTN/ISDN using ISDN Primary Rate Interface i.e. 30B+D. 3.16.2 Whenever any extension goes off-hook, the PABX shall feed dial tone to its extensions. 3.16.3 Depending on the requirements of the PABX, the exchange will send all/last few digits of called number to the PABX. The PABX shall indicate the end of digits by sending appropriate signal to the main exchange. 3.16.4 Access to Public Network: The Public Network shall be accessed by dialing single digit. Dialling of `0' (or any other suitable code) shall be used to provide access to the public network (O/G junctions) and the junction access should have a random or rotating priority to distribute the traffic. 3.16.5 The PABX shall provide for time out and release of outgoing junction/circuit under held up condition during pre-dialling, dialling, post-dialling, ringing phase and CSH condition. 3.16.6 Calling Line Identification 3.16.6.1 For all types of calls from public network, Calling Line Identification (CLI) of a pre-defined number of digits will be sent by the main exchange. At present it is 10 digits CLI, but it can be increased up to 16 digits. The PABX shall be able to receive complete Calling Line Identification (CLI), (consisting of country code + Area code + Directory number) from the main exchange (at present country code in not included in CLI). It shall not be possible to change the CLI in PABX. 3.16.6.2 For all types of calls made from extension to public network (If extension numbering scheme used is from ‘group of directory numbers’ allotted by service provider), CLI of extension number shall be send to the main
GR No. TEC/GR/SW/PBX-001/02/OCT-17 30
exchange.
3.16.7 Direct Inward Dialling (DID) Facility The PABX shall support the facility where any Public network subscriber shall be able to dial the PABX extension number without the assistance of any PABX operator/attendant. This facility can be implemented by any one of the following methods: i. Direct Inward Dialling (DID) : Service provider shall allot ‘a group of directory numbers’ from the concerned parent exchange to the PABX. All extension users will have directory numbers as per E.164 numbering plan of the service provider. In such cases, main exchange shall send digits of extension number to PABX. ii. Each PABX shall be allotted a single directory number by the service provider. When a Public network subscriber dials PABX number, PABX shall feed announcement to caller and calling subscriber shall dial the extension number. 3.16.8 Alarm/Fault Diagnosis of Interface For PRI interface, PABX shall support display of status of PRI towards the main exchange and shall generate the following alarms. a) Loss of the incoming signal at 2048 kbit/s. b) Loss of frame alignment. – PABX should detect ‘loss of frame alignment’ within 3 ms. c) Loss of Multiframe alignment (Multiframe alignment is assumed to have been lost when, for a period of one or two multiframes, all the bits in time slot 16 are in state 0. Multiframe alignment is assumed to have been recovered only when at least one bit in state 1 is present in the time slot 16) d) Excessive bit error ratio.
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SECTION- G (For Type-6 Interface only) 3.17 V 5.2 Interface In addition to functional requirements described in section A, this section specifies the functional requirements for PABX with Type-6 interface (E1 i.e. 2048 kbits/s PCM junction) both for outgoing calls from PABX to Public network and incoming calls from the Public networks using V5.2 signalling. 3.17.1 The PABX shall have the capability to interconnect with PSTN using E1 i.e. 2048 kbit/s PCM junctions. 3.17.2 Whenever any extension goes off-hook, the PABX shall feed dial tone to its extensions. If a second dial tone is required to dial public network, it shall be fed after ensuring that a free junction is available to the main exchange. 3.17.3 Depending on the requirement of the PABX, the exchange will send all/last few digits of called number to the PABX. 3.17.4 Access to Public Network: The Public Network shall be accessed by dialing single digit. Dialling of `0’ shall be used to provide access to the PSTN network (O/G junctions) and the junction access should have a random or rotating priority to distribute the traffic. 3.17.5 The PABX shall provide for time out and release of outgoing junction/circuit under held up condition during pre-dialling, dialing, post-dialling, ringing phase and CSH condition. 3.17.6 Direct Inward Dialling (DID) Facility The PABX shall support the facility where any Public network subscriber shall be able to dial the PABX extension number without the assistance of any PABX operator/attendant. This facility can be implemented by any one of the following methods: i. Direct Inward Dialling (DID): Service provider shall allot ‘a group of directory numbers’ from the concerned parent exchange to the PABX. All extension users will have directory numbers as per E.164 numbering plan of the service provider. In such cases, main exchange shall send digits of extension number to PABX. ii. Each PABX shall be allotted a single directory number by the service provider. When a Public network subscriber dials PABX number,
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PABX shall feed announcement to caller and calling subscriber shall dial the extension number. 3.17.7 Clock: PABX shall extract clock from 2048 kbits/s link between the main exchange and the PABX and shall use it for the timing purpose. 3.17.8 For all types of calls to PSTN, Calling Line Identification (CLI) of a pre- defined number of digits will be requested by the main exchange. At present it is 10 digits CLI, but it can be increased up to 16 digits. The PABX shall be able to send complete Calling Line Identification (CLI), (consisting of country code + Area code + Extension number including the main exchange level) on demand to the PSTN exchange (at present country code is not included in CLI). It shall not be possible to change the CLI in PABX. In case CLI is not sent by the PABX, the call shall be rejected by the main exchange.
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SECTION- H (For Type-7 Interface only) 3.18 Ethernet based Interface In addition to functional requirements described in section A, this section specifies the functional requirements for PABX with Type-7 interface using SIP or H.323 signaling
3.18.1 Addressing: The PABX shall support IP addressing as per version 4 (IPv4) as per IETF RFC 791 as well as to version 6 (IPv6) as per IETF RFC 2460. 3.18.2 PABX shall have the capability to inter-work with IP networks supporting IPv4 as per IETF RFC 791 as well IPv6 as per IETF RFC 2460. 3.18.3 Failures in the PABX causing inability to receive incoming calls shall be signalled from the PABX to the public exchange/ network by suitable means. 3.18.4 The PABX shall have the capability to interconnect with any switching Node (MediaGateway or Soft-switch or CSCF) 3.18.5 The physical interfaces (Ethernet connections used) shall consist of 10/100/1000 over twisted pair or coaxial or Fiber (as per IEEE, 802.3u/ IEEE 802.3z). Category 5 cables/ Category 6 cable/8P8C modular connector shall be used
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4.0 Interface Requirement
4.1 The equipment shall provide interoperability with all types of exchanges in the present Indian Telecom Network.
4.2 The equipment shall not necessitate any changes in the existing Public Telecom Network and shall not impact the interfaces and protocols in these networks.
4.3 Depending on the capacity of the PABX, the exchange will send all/last few digits of called number (as per requirements) to the PABX. The PABX shall indicate the end of digits by sending appropriate signal to the main exchange.
4.4 Failures in the PABX causing inability to receive incoming calls should be signalled from the PABX to the public exchange by suitable means.
4.5 Signalling between PABX and main exchange:
Applicable only for type(s) of interface(s) for which applicant has sought type approval.
4.5.1 2 W DEL Interface (Applicable for Type-1 Interface only) For outgoing calls from PABX to the main exchange, PABX shall use DTMF signalling. Applicant may apply for decadic signalling also. If applicant applies for decadic signalling, same shall be indicated on the TAC, after testing. (a) In case of decadic dialling out by PABX on the junctions, decadic pulses shall have the following characteristics (Applicable only if the applicant has applied for decadic signalling): i Dial speed shall be 10 ± 0.5 IPS. ii Make/break ratio shall be 1:2 nominal with break period between 60% to 70% iii IDP of 800 msec. ± 10% (b) In case of DTMF dialling out by the PABX, the specifications for the same shall be in conformity with ITU-T recommendation Q.23. The level of the signals shall be as follows: Low group -8 ± 2 dBm High-group -6 ± 2 dBm
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Pre-emphasis 2 ± 1 dB (c) Transmission of DTMF Signalling shall be as per ITU-T recommendation Q.23 and reception of DTMF Signalling shall be as per ITU-T recommendation Q.24. The number of digits/characters used for Signalling shall be 12 and a set of two frequencies shall be used for each digit/character. Low frequency group : 697,770, 852, 941 Hz High frequency group : 1209, 1336, 1477 Hz
Digit Frequencies Digit Frequencies used used Digit 1 697 & 1209 digit 7 852 & 1209 digit 2 697 & 1336 digit 8 852 & 1336 digit 3 697 & 1477 digit 9 852 & 1477 digit 4 770 & 1209 digit 0 941 & 1336 digit 5 770 & 1336 * 941 & 1209 digit 6 770 & 1477 # 941 & 1477 (d) Each transmitted Signalling shall be within ± 1.8% of the nominal frequency. (e) The total distortion products (resulting from harmonic or inter-modulation) must be at least 20 dB below normal frequencies. (f) Level difference between any two frequencies forming a signal should not be more than 2 dB ± 1 dB such that the higher frequency component of the signal is of higher level. (g) During DTMF Signalling, the d.c. loop shall be maintained. (h) The PABX shall not provide for more than one automatic repeat dialling on the junction line. In case of DTMF redial, the on/off period shall be at least 40/40 msecs.
4.5.2 E1 R2 Interface (Applicable for Type-2 Interface only) (a) The PABX shall feed dial tone to its extensions. If a second dial tone is required to dial public networks, it shall be fed by the PABX after ensuring that a free circuit is available to the main exchange. (b) The signalling between the PABX and PSTN networks shall be as follows:
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(i) Line Signalling: Line signals are used for seizure, supervision and disconnection of communication. Line signalling of digital type 1 or type 2 or type 3 as described below shall be used. a. Digital Type 1 Signalling Operating condition Forward Backward af bf cf ab bb Idle 1 0 0 1 0 Seizure 0 0 0 1 0 Acknowledgement 0 0 0 1 1 Answer 0 0 0 0 1 Metering Pulse 0 0 0 1 1 (180 - 270 ms) Clear back 0 0 0 1 1 Clear forward 1 0 0 0 or 1 1 Release guard 1 0 0 1 0 Trunk Offering and Re-ringing a) TKO press key 0 0 1 1 1 b) False answer 0 0 1 0 1 c) Release key 0 0 0 0 1 d) “B” party on 0 0 0 1 1 hook e) Re-verify 0 0 1 1 1 Blocking 1 0 0 1 1 Notes: 1. For all supervisory signals bf = 0; a change to bf = 1 indicates a fault. 2. df, cb, db are spare bits, df =db = 1, and cb = 0, are assigned according to ITU-T Recommendation G.732. b. Digital Type 2
Signalling Operating condition Forward Backward
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af bf a b b b Idle 1 1 1 0 Seizure 0 1 1 0 Acknowledgement 0 1 1 1 Answer 0 1 0 1 Metering Pulse 0 1 1 1 Clear back 0 1 1 1 Clear forward 1 1 0 or 1 1 Release guard 1 1 1 0 Trunk Offering and Re- ringing a) TKO press key 0 0 1 1 b) False answer 0 0 0 1 c) Release key 0 1 0 1 d) “B” party on hook 0 1 1 1 e) Re-verify 0 0 1 1 Blocking 1 1 1 1
Forward cf = 0 Backward cb = 0 df = 1 db = 1
C. Digital Type-3
Signalling Operating condition Forward Backward af bf a b b b Idle 1 1 1 1 Seizure 0 1 1 1 Answer 0 1 0 1 Metering Pulse 0 1 1 1 Clear back 0 1 1 1
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Clear forward 1 1 0 or 1 1 Trunk Offering and Re-ringing a) TKO press key 0 0 1 1 b) False answer 0 0 0 1 c) Release key 0 1 0 1 d) “B” party on hook 0 1 1 1 e) Re-verify 0 0 1 1 Blocking 1 1 0 1
(ii) Register Signalling: The register signalling between the main exchange and the PABX shall be Indian R2 modified MFC signalling as described below: Digital exchanges shall provide interface on 2048 kbit/s PCMs. a. The frequencies used in the backward direction are 660, 780, 900, 1020 and 1140 Hz. Those used in forward direction are 1380, 1500, 1620, 1740, and 1860 Hz. MF signals shall be sent in the form of pulses of nominal duration 150 ms b. Forward signals- The forward frequencies shall be used to send digits (when used as Group-I forward signals) or to send the category of calling subscriber (when used as group-II forward signals). c. Backward signals - The backward frequencies shall be used to make further demands (when used as Group-A backward signals), or to report back the called line condition to the calling side (when used as Group-B backward signals). In electronic exchanges all 5 backward frequencies shall be equipped. d. The significance of the forward signals and backward signals, as interpreted in the Indian network, are given in tables below: Group I - Forward Signals: Signals Indian National MFC scheme I.1 Digit 1 I.2 Digit 2 I.3 Digit 3 I.4 Digit 4 I.5 Digit 5
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I.6 Digit 6 I.7 Digit 7 I.8 Digit 8 I.9 Digit 9 I.10 Digit 0
Group II - Forward Signals:
Signals Indian National MFC scheme
II.1 Ordinary Subscriber II.2 Priority Subscriber II.3 Spare at present (proposed for use in future for ‘maintenance equipment call’) II.4 Spare at present II.5 Operator II.6 STD coin box. II.7 Spare II.8 Spare
II.9 Spare II.10 Spare
Group A - Backward Signals:
Signals Indian National MFC scheme A.1 Send next digit (n+1) A.2 Restart A.3 Change over to reception of B-signals A.4 Calling line identification-see note A.5 Send category of the calling subscriber A.6 Switch-through the speech path A.7 Send last but two digit (n-2) Not available in A.8 Send last but three digit (n-3) local exchange A.9 Send last but one digit (n-1) at present A.10 Spare at present (may be used for trunk congestion if network
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permits)
Group B - Backward Signals
Signals Indian National MFC scheme B.1 Spare B.2 Changed number B.3 Called line busy B.4 Congestion B.5 Unallotted number B.6 Normal subscriber, free, with metering B.7 Spare B.8 Spare; not available in local exchange B.9 Spare; not available in local exchange B.10 Spare; not available in local exchange e. For CLI, requesting exchange/PABX (called side) shall send A4 backward signal. The calling side on receipt of A4 backward signal, shall send forward signal ‘1’. On receipt of forward signal ‘1’, the requesting side shall send A4 signal and the calling side shall send first digit of CLI. Similarly, all digits of CLI shall be received by the requesting side. f. Sequence of MF signaling shall as per ITU-T Recommendations Q.440, Q.441 and Q.442. (c) Depending on the capacity of the PABX, the exchange will send 3 to 4 digits to the PABX. The PABX shall indicate the end of digits, by sending ‘A3’ and ‘B’ group signal to the main exchange after required digits are received. (d) In case of Register signalling between the main exchange and PABX, B7 and any other spare signal shall not be used for the inter-working.
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4.5.3 BRI at S-Interface (Applicable for Type-3 Interface only)
4.5.3.1 Signalling
(a) Signalling shall be as per ITU-T Recommendation I.430- Basic user- network interface - Layer 1 specification. (b) Layer 2 Signalling shall be as per ITU-T Recommendation Q.920 (03/93)-ISDN user-network interface data link Layer-General aspects, Q.921 (09/97) ISDN user-network interface - Data link layer specification. (c) Layer 3 message structure and coding of information elements shall be as per ITU-T Recommendation Q.931.
4.5.4 BRI at U-Interface (Applicable for Type-4 Interface only) 4.5.4.1 Signalling (a) Layer 1 Signalling shall be as per ITU-T Recommendation I.430- Basic user-network interface - Layer 1 specification. (b) Layer 2 Signalling shall be as per ITU-T Recommendation Q.920 (03/93)-ISDN user-network interface data link Layer-General aspects, Q.921 (09/97) ISDN user-network interface - Data link layer specification. (c) Layer 3 message structure and coding of information elements shall be as per ITU-T Recommendation Q.931.
4.5.5 PRI Interface (Applicable for Type-5 Interface only) 4.5.5.1 Signalling
(a) Layer 1 requirements shall be as per ITU-T- Primary Rate User- Network Interface – Layer 1 Specification.
The primary rate shall support only the point-to-point configuration. Point-to-point configuration at layer 1 implies that for each direction only one source (transmitter) and one sink (receiver) are connected to the interface.
i). Nominal bit rate: 2048 kbit/s.
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ii). Bit rate accuracy: ±50 ppm (±102.4 bit/s). iii). Code: High density bipolar of order 3 (HDB3)
(b) Layer 2 requirements shall be as per ITU-T Q.920 - ISDN user-network interface data link Layer-General aspects and Q.921 - ISDN user- network interface - Data link layer specifications (c) Layer 3 message structure and coding of information elements shall be as per ITU-T Recommendation Q.931.
4.5.6 V 5.2 Interface (Applicable for Type-6 Interface only) 4.5.6.1 Signalling
Signalling shall be as per National standard for V 5.2 Interface No. TEC/SD/SW/VAN-SIG.
4.5.7 Ethernet based Interface (Applicable for Type-7 Interface only) 4.5.7.1 Signalling
(a) At Layer 2, Ethernet 802.1p/Q standards define the bit markings of Ethernet packet header which are used to prioritize packets at Layer 2. (b) At Layer 3, IP standard DiffServ defines bit markings in the Type-of- Service (TOS) fields in the IP header, which will identify a packet to be associated with a specific service. On IP equipment end-to-end, these services can be administered. (c) Ethernet interface shall support standard based protocol (RTCP) for voice quality monitoring. (d) Signalling between PABX and IP Core Network shall comply the following RFCs. S.No. Document No. Title 1 IETF RFC 2460 IPv6 addressing 2 IETF RFC 3261 SIP: Session Initiation Protocol 3 IETF RFC 4733/2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals 4 IETF RFC 3550 Real Time Transport Protocol (RTP)
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5 IETF RFC 793 or Transmission Control Protocol (TCP)/ IETF RFC 768 User Datagram Protocol (UDP)
(e) If Applicant has sought approval for H.323 based IP Trunk interface, signaling shall be as per ITU-T Recommendation H.323 (12/2009) with Amendment 1 (03/2013) {Latest version shall be applicable} (f) The system shall support Network Time Protocol as per RFC 1305 to synchronies the system date and time with network devices. 4.5.7.2 Codec (a) The interface shall support the procedures for Codec negotiation, in association with the Main Switching Node. It shall support G.711 and G.729 codec as per ITU-T recommendations and shall cater for at least 3.4 KHz bandwidth. For video, it shall support at least H.264. Applicant may apply for additional codec supported and the same may be mentioned in the approval certificate after testing. (b) The Codec modification/negotiation procedures shall also be provided, wherein the Codec selected for a call can be modified in any direction and at any time during the active phase of the call. (c) It shall be possible to convey the list of available codec that can be used for the call set-up. (d) Any voice processing functions shall not interfere with transparent interchange of FAX signals. It shall support FAX over IP as per ITU-T Recommendation T.38.
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5.0 Quality Requirements
5.1 In case of Ethernet interface (i.e. type-7 interface), PABX should support RTCP-XR protocol for monitoring real time VoIP voice quality.
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6.0 EMI/EMC Requirements
The equipment shall conform to the following EMC requirements for Class A: General Electromagnetic Compatibility (EMC) Requirements: - The equipment shall conform to the EMC requirements as per the following standards and limits indicated therein. A test certificate and test report shall be furnished from an accredited test agency.
a) Conducted and radiated emission(applicable to telecom equipment): Name of EMC Standard: “CISPR 22 (2008) - Limits and methods of measurement of radio disturbance characteristics of Information Technology Equipment". Limits: -
i. To comply with Class B OR Class A of CISPR 22 (2008). ii. The values of limits shall be as per TEC Standard No. TEC/SD/DD/EMC-221/05/OCT-16.
iii. For Radiated Emission tests, limits below 1 GHz shall be as per Table 4 (a) or 5 (a) for measuring distance of 10m OR Table 4 (a1) or 5 (a1) for measuring distance of 3m. OR Conducted and radiated emission (applicable to instruments such as power meter, frequency counter etc.): Name of EMC Standard: "CISPR 11 {2015}- Industrial, scientific and medical (ISM) radio- frequency Equipment-Electromagnetic disturbance characteristics- Limits and methods of measurement" Limits: - i. To comply with the category of Group 1 of Class A of CISPR 11 {2015} ii. The values of limits shall be as per clause No. 8.5.2 of TEC Standard No.TEC/SD/DD/EMC-221/05/OCT-16.
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b) Immunity to Electrostatic discharge: Name of EMC Standard: IEC 61000-4-2 {2008) "Testing and measurement techniques of Electrostatic discharge immunity test". Limits: - i. Contact discharge level 2 {± 4 kV} or higher voltage; ii. Air discharge level 3 {± 8 kV} or higher voltage;
c) Immunity to radiated RF: Name of EMC Standard: IEC 61000-4-3 (2010) "Testing and measurement techniques-Radiated RF Electromagnetic Field Immunity test" Limits:-
For Telecom Equipment and Telecom Terminal Equipment with Voice
interface (s)
i. Under Test level 2 {Test field strength of 3 V/m} for general purposes in frequency range 80 MHz to 1000 MHz and ii. Under test level 3 (10 V/m) for protection against digital radio telephones and other RF devices in frequency ranges 800 MHz to 960 MHz and 1.4 GHz to 6.0 GHz.
For Telecom Terminal Equipment without Voice interface (s)
Under Test level 2 {Test field strength of 3 V/m} for general purposes in frequency range 80 MHz to 1000 MHz and for protection against digital radio telephones and other RF devices in frequency ranges 800 MHz to 960 MHz and 1.4 GHz to 6.0 GHz.
d) Immunity to fast transients (burst):
Name of EMC Standard:IEC 61000- 4- 4 {2012) "Testing and measurement techniques of electrical fast transients/burst immunity test"
Limits:-
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Test Level 2 i.e. a) 1 kV for AC/DC power lines; b) 0. 5 kV for signal / control / data / telecom lines;
e) Immunity to surges: Name of EMC Standard: IEC 61000-4-5 (2014) “Testing & Measurement techniques for Surge immunity test" Limits:- i. For mains power input ports: (a)2 kV peak open circuit voltage for line to ground coupling (b) 1 kV peak open circuit voltage for line to line coupling ii. For telecom ports: (a) 2 kV peak open circuit voltage for line to ground (b)2 kV peak open circuit voltage for line to line coupling.
f) Immunity to conducted disturbance induced by Radio frequency fields: Name of EMC Standard: IEC 61000-4-6 (2013) "Testing & measurement Techniques-Immunity to conducted disturbances induced by radio- frequency fields” Limits:- Under the test level 2 {3 V r.m.s.}in the frequency range 150 kHz-80 MHz for AC / DC lines and Signal /Control/telecom lines.
g) Immunity to voltage dips & short interruptions (applicable to only ac mains power input ports, if any): Name of EMC Standard: IEC 61000-4-11 (2004) “Testing & measurement techniques- voltage dips, short interruptions and voltage variations immunity tests” Limits:- i. a voltage dip corresponding to a reduction of the supply voltage of 30% for 500ms (i.e. 70 % supply voltage for 500ms) ii. a voltage dip corresponding to a reduction of the supply voltage of 60% for 200ms; (i.e. 40% supply voltage for 200ms) iii. a voltage interruption corresponding to a reduction of supply voltage of > 95% for 5s.
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iv. a voltage interruption corresponding to a reduction of supply voltage of >95% for 10ms.
Note 1: Classification of the equipment: Class B: Class B is a category of apparatus which satisfies the class B disturbance limits. Class B is intended primarily for use in the domestic environment and may include: • Equipment with no fixed place of use; for example, portable equipment powered by built in batteries; • Telecommunication terminal equipment powered by the telecommunication networks • Personal computers and auxiliary connected equipment.
Please note that the domestic environment is an environment where the use of broadcast radio and television receivers may be expected within a distance of 10 m of the apparatus connected.
Class A: Class A is a category of all other equipment, which satisfies the class A limits but not the class B limits.
Note 2: The test agency for EMC tests shall be an accredited agency and details of accreditation shall be submitted.
Note 3: For checking compliance with the above EMC requirements, the method of measurements shall be in accordance with TEC Standard No. TEC/SD/DD/EMC-221/05/OCT-16 and the references mentioned therein unless otherwise specified specifically. Alternatively, corresponding relevant Euro Norms of the above IEC/CISPR standards are also acceptable subject to the condition that frequency range and test level are met as per above mentioned sub clauses (a) to (g) and TEC Standard No. TEC/SD/DD/EMC- 221/05/OCT-16. The details of IEC/CISPR and their corresponding Euro
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Norms are as follows:
IEC/CISPR Euro Norm CISPR 11 EN 55011 CISPR 22 EN 55022 IEC 61000-4-2 EN 61000-4-2 IEC 61000-4-3 EN 61000-4-3 IEC 61000-4-4 EN 61000-4-4 IEC 61000-4-5 EN 61000-4-5 IEC 61000-4-6 EN 61000-4-6 IEC 61000-4-11 EN 61000-4-11
The manufacturer / supplier shall submit a test certificate and test report for EMI/EMC compliance from test agency. The test agency for EMI/EMC tests shall be an accredited agency and details of accreditation shall be submitted.
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7.0 Safety Requirements
7.1 The equipment shall conform to IS 13252 part 1:2010- “Information Technology Equipment – Safety- Part 1: General Requirements” [equivalent to IEC 60950-1 {2005} “Information Technology Equipment –Safety- Part 1: General Requirements” and 7.2 A test certificate and test report shall be furnished from an accredited test agency. 7.3 The test agency for safety requirements tests shall be an ISO 17025 accredited agency and details of accreditation shall be submitted.
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8.0 Security Requirements 8.1. 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.
8.2. Security measures such as user account, user’s password, equipment authentication & registration information shall be transmitted in the secured form. (Applicable only in case of Type-7 interface)
8.3. System Access:
i Suitable safeguards shall be provided in the programs to debar unauthorised persons from making any changes in the memory contents or office data. Access to system operations shall be controlled through multi-level password and authentication checks. ii The man-machine communication programs shall have the facility of restricting the use of certain commands or procedures to certain passwords. iii Operator/Maintenance console shall be GUI based. Necessary software package to prevent loading of any unauthorised software or driver on the I/O terminal (without specific authorisation from PC administrator) shall be provided. iv The system must support ‘session logout timing with configurable time periods. v The system must support password encryption. 8.4 In case of Ethernet interface (i.e. type-7 interface), In case of Ethernet interface (i.e. type-7 interface), PABX should support SRTP protocol for encryption of voice, video and DTMF deliverd by RFC2833/4733
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CHAPTER - 2
9.0 Information for the procurer of Product This chapter describes the requirements, which are useful for efficient functioning of the system. However, the purchase will finalise these requiremets at the time of procurement.
9.1 General
9.1.1 The system architecture shall be so designed that it enables expansion of the system without any change in software.
9.1.2 The system software should be modular so that any functionality can added or removed without disturbing the other functionalities.
9.1.3 The up-gradation should be possible by simply adding peripheral/interface/line cards on the same set of control cards & CPU without compromising on any function/feature of the system or any degradation of service.
9.1.4 The system should support distinctive ringing for external calls (i.e. calls from public network), intra PABX calls and diverted calls.
9.2 System Redundancy:
9.2.1. The system should be provided with 100% duplicated Control Unit in Hot standby mode.
9.2.2. Redundancy shall be provided for the following subsystem of PABX:
i Common Control ii Switching Network (in case TDM switching is used) iii Tones iv Main and standby memory
9.2.3. The system should provide complete set of control cards in redundant mode. In case of failure of one control card, the duplicated card should take over immediately with the current database, on which the system is working, without disconnection of established calls.
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9.2.4. The offered system should be capable of Hot Swapping of all control cards without switching off the system, where the necessary cards can be interchanged or replaced even in online conditions without disruption of services.
9.2.5. It should be possible to reach the ultimate capacity of the PABX without any up gradation of the Control cards and need of additional Control cards.
9.2.6. The system shall adopt active/standby mode, load sharing and redundancy configuration for the boards and optimizes fault detection and isolation techniques of the boards and the system to improve the maintainability of the whole system.
9.2.7. System overload control shall provide overload restrictions, dynamic code adjustment mode and traffic control to fully ensure the reliability of the system.
9.2.8. The system should support the concept of Survivable Branch Gateway. In this the branch gateway communicated to main site over WAN. In case WAN is down, the users at branch gateway are server locally by this gateway. SO the call switching happens locally when WAN is down. (Applicable for type -7 interface only).
9.3 System Security: -
a. The System must support Syslog services for command and configuration control accounting with a minimum of 5-day history. b. Internal OS controls for remote point of access restriction and service availability c. Account access authentication/restriction at different levels shall be provided so as to prevent unauthorized access or interference to services, calls, protocol and data.
9.4 Maintenance Console:
a. The offered system should be provided with a PC based and software up-gradable maintenance console.
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b. The operational functions like creating/deleting an extension number, allocating, changing and removing any fuser’s facility on any extension number.
c. The system should support fault diagnosis up to card level.
d. Call statistic reports as required should be available on the monitor as well as printouts. The formats of printouts should be modifiable.
9.5 Operator Console:
Operator console shall support the following features:
i The Operator shall be able to identify the category of an incoming call e.g. internal, external, diverted, networked
ii The operator shall be able to transfer incoming calls to another number, internal or external.
iii. The operator shall be able to camp on a call on to a busy extension.
iv. The operator shall be able to make an internal/external call and shall be able to transfer it another number.
v. The operator shall be able to park calls and retrieve them later.
vi. An extension shall be able to dial the operator by a general access code (say 9) and by an individual extension number.
vii. Busy Lamp Field – to indicate the busy/ free status of the defined extensions and trunks groups.
9.6 PC for Operator Console & Maintenance Console- Following are the suggestive specifications for PC for Operator/ maintenance console:
a. Windows XP or higher with Ms office 2007 professional b. Intel Pentium 4 Processor, 2.9 GHz or above c. HDD – 100 GB d. RAM - 512 MB DDR RAM (upgradeable up to 1 GB)
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e. Ethernet Interface (10/100/1000Mbps) f. USB Ports g. 1 Serial and 1 parallel Port h. FDD -1.44’ i. 52 X CD ROM/DVD with Multimedia j. Keyboard k. Mouse l. Flat Colour Monitor (Minimum size 17”)
9.7 Auto-attendant
This is the part of the system that initially answers the incoming calls, eliminating the need for a full-time receptionist. Callers hear a custom business greeting, and then are given options for routing their calls.
Virtual PBX gives many options for call routing, including:
A specific extension number. An ACD queue (e.g. dial 1 for sales, 2 for support etc.) Select an employee by first or last name from a company directory Send a fax to an employee extension Enter a tele-conference Transfer to an information extension, such as a message giving driving directions or office hours It should be possible to customize the response of the auto-attendant through one’s own custom greeting.
9.8 Diagnostic programs to localise hardware faults
i. On a faulty condition, the software shall provide for isolating the faulty subsystem and then automatically activating the diagnostic programs for diagnostic purpose to localize fault up to the card level.
9.9 System back-up
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a. It should be possible to save system back-up automatically/ manually in the system hard disk. In addition to it should also be possible to save system back-up in CD-ROM/ DVD/Pen drive/ Optical disk or Cartridge etc.
b. It shall be possible to take complete system backup on the same device. It shall also be possible to load the system from the backup. It shall be possible to store multiple files of charging information, detailed billing information, traffic statistics, command log, system software, office data etc. on the same device.
9.10 Power Supply
Option 1:
The equipment shall be capable of working with –40 V to –57 V. D .C. input from power supply.
Switching mode Power Supply (SMPS) and VRLA battery to be used shall be as per TEC Generic Requirements No. GR/SMP –01 and GR/BAT-01 Respectively. Power supply and battery shall be modular and expendable to support the ultimate equipment configuration.
Option 2:
AC Mains supply of 220 Volts with a tolerance of -15% to + 10% would be available. The frequency may be 50 Hz + 2 Hz. UPS and other power requirements are to be specified by the system developer. Relevant TEC Specification/ GR as applicable may be referred
Purchase may specify the power requirement as per option 1 or 2.
9.11 Day and night modes
PABX system acts differently after business hours. This feature allows the system to route incoming calls depending on the current time of day. One may have several after-hours options, available at the click of a button:
a. Change the greeting callers hear, but leave the system fully active. Callers will know that business hours are over, but may still try to connect to an extension. With follow-me calling, answering options and
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voicemail each extension can decide if he wants to continue to take calls or simply collect messages.
b. Change the greeting and send all calls to voicemail. Called umber shall know that messages have been recorded through notification features.
c. Leave the system in full operation, or put it back when the office opens again.
9.12 User Facilities
PABX shall provide the following facilities/services for all types of extensions:
9.12.1 O/G Call Restriction – Administrative controlled
Facility shall be provided (with suitable safeguards), to enable authorised
operating personnel to allow/bar any extension any facility/services. It shall
be possible to administratively assign the following access restrictions:
i Calls to public network barred ii NSD barred iii ISD barred iv Any particular level (say level 9) v Premium rate services
9.12.2 O/G Call Restriction – Subscriber controlled (For access to public network, NSD barring, ISD barring)
This feature will enable a subscriber to bar certain O/G directions from his line. The types of barred directions will be prescribed by administration and the subscriber should be able to select, activate and deactivate them. The procedure for initial password registration and modification of password, activation/deactivation of any restriction shall be defined clearly in the user manual.
9.12.3 Automatic Alarm Call service (Wake-up-call): This service is also called Morning Alarm Service.
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Feature Description:
i. This feature allows a subscriber to get an alarm call on his telephone after activating this facility by a standard procedure. ii. At the specified time a call is made by the system to the subscriber and a suitable announcement is fed to subscriber, who has registered the alarm. iii. If the subscriber does not answer the phone within one minute (or ringing time out whichever is later) or if the subscriber is busy, the alarm call will be repeated after a pre-defied time. iv. If the repeat attempt is also unsuccessful, request shall be cancelled. v. The subscriber can also himself cancel the request through a pre- specified procedure. vi. The subscriber, however, may have the possibility to register more than one alarm call (at least 3) to be made to his line with in a 24-hour period.
9.12.3.1 The number of Wake-up-call registrations for a particular wake-up alarm time shall be defined by the tendering authority (suggested fig. is 1% of the equipped capacity).
9.12.3.2 The procedure for alarm registration/cancellation shall be defined clearly in the user manual.
9.12.4 Call Waiting
Feature Description
a. A subscriber 'A' engaged on an existing call with subscriber ‘B’ is given an indication (Call Waiting Tone) that a caller subscriber ‘C’ is attempting to obtain connection. ‘C’ party shall be fed an announcement to indicate that ‘A’ party is busy and he is in the queue. b. Subscriber 'A' may answer by replacing the handset (acceptance by clearing). In case the ‘A’ party releases the call to ‘B’, ring back tone shall be fed to ‘C’ party after stopping the announcement and send ring to subscriber ‘A’
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c. Alternatively, he may ignore or reject the indication and continue with the existing call. d. Terminate the existing call and answer subscriber 'C'. e. Hold the existing call and answer subscriber 'C' (acceptance without clearing).
9.12.5 Absent subscriber Service:
i. The feature will permit a subscriber to enable/disable diversion of all incoming calls automatically to an announcement. ii. During the service being operative, the subscriber should be given an indication in the form of a special dial tone. iii. The subscriber should be able to originate O/G calls.
9.12.6 Do Not Disturb
i Do not disturb service is used by the subscribers who do not wish to be disturbed by incoming calls for a certain period of time. After activation of this service, the incoming calls are diverted to an answering machine/Announcement ii Outgoing calls shall normally be possible. iii Alarm calls shall be sent to subscriber. iv When the service is activated a special dial tone shall be given for outgoing calls reminding the service.
v This is a subscriber controlled service.
9.12.7 Call Forwarding on Busy, No Reply, Immediate
a. Access restrictions (as per subscriber controlled Public network/STD) shall be applicable for registration of call forwarding only and not when call forwarding takes place i.e. Registration of call forwarding for Public network/STD shall not be possible if access to Public network/STD is restricted. Once registered, forwarding of calls to Public network/STD shall be possible even access to Public network/ STD is restricted by
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the calling subscriber. b. Call forwarding under Administration control shall be kept under higher password level. c. It shall be possible to define ringing duration for call forwarding on “no reply”. Normally, the system default timer of ringing duration shall be 30 seconds d. After analysis of part of called number, it shall be possible to either allow or restrict “Call Forwarding”. It shall be possible to restrict call forwarding on STD number or `premium rate service’. e. If “Call Forwarding” is restricted based on the above analysis, this will override any permission allowed to individual subscribers. f. If “Call Forwarding” is allowed based on the above analysis, the permission/ restriction allowed to individual subscribers shall also apply. g. It shall be possible to define time dependent call forwarding. h. When both the services, namely, “call waiting” and “call forwarding on busy” are active, “call forwarding on busy” shall take precedence.
9.12.8 Incoming only line
a By this facility a subscriber can receive ONLY incoming call made to him. He can NOT originate calls from his line.
b In case an originating call is attempted, suitable announcement shall be fed to calling subscriber indicating that this facility is not available on this line.
c Originating calls to emergency numbers (e.g. 100, 101, 102 and PABX operator) may normally be allowed from this number.
d This is an Administrative Controlled feature.
9.12.9 Outgoing only lines
i By this facility a subscriber is allowed to make ONLY originating calls from his line. Incoming calls to this subscriber are NOT allowed.
ii Automatic calls (Morning alarm, calls from PABX operator) may normally be allowed to this number.
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This is an Administrative Controlled feature.
9.12.10 Audio Conference Call:
Switch from one call to the other as required, secrecy being provided between the two calls.
Introduce a common speech path so that all the subscribers in conference are able to speak to each other
Various numbers in the conference can be either internal extension or external number in public network. Maximum number of subscribers in a conference and maximum number of simultaneous multiparty audio conferencing shall be defined by the tendering authority
9.12.11 Fixed Destination Call - Immediate or Timed out (Hot line)
i This feature will enable the subscriber to set up a call to a predetermined number by lifting handset only.
ii When the subscriber lifts the handset or off-hook condition is given by automatic equipment, the exchange sets up immediately or after a delay period of some predefined time (say 5 secs.) a connection to a pre-determined telephone number.
iii Within the delay period the subscriber will be able to initiate a normal call.
iv This telephone number may be intra-PABX, local, national, international, or the number of an operator service.
v The number registration can be done either by the subscriber or on request, by the administration.
vi In case of hot line (Timed out) the access check (as per Dynamic STD/ISD) shall be done at the time of registration by the subscriber.
vii Administration controlled Hot Line should be kept at higher level of password.
9.12.12 Call Transfer
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After answering an Incoming call, any extension subscriber shall be able to transfer the call to another subscriber of the PABX, without assistance of the operator.
9.12.13 Call Forwarding to Fixed Number
i In addition to call forwarding services, it shall be possible to forward the call to fixed number definable by MML commands (e.g. Voice Mail Service) for all three conditions i.e. busy, no reply and Immediate. ii It shall be possible for a subscriber to activate and deactivate this service by dialling a service code definable by MML. iii For a particular condition, call forwarding to a variable number shall have precedence. 9.12.14 Automatic Call Back System shall support call back to the phone number from which the message was received. The subscriber should be able to select, activate and deactivate the above service. 9.12.15 Call Completion on Busy Subscriber (CCBS):
CCBS service enables a calling user A, upon encountering a busy destination B, to be notified when the busy destination B becomes free and to have the service provider reinitiate the call to the specified destination B if user A desires.
i The service shall be allowed only when authorised by Administration.
ii The Activation and Deactivation codes for CCBS feature shall be programmable.
iii The sequence would be as follows:
a. User A calls destination B. b. Destination B is busy. c. User A has subscribed to the CCBS service and the network notifies him that the service can be invoked. d. User A invokes the CCBS service. e. The network will monitor for destination B becoming free.
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f. When destination B becomes free, then the network will wait a short time as defined in the destination B idle guard timer in order to allow the resources to be reused for originating a call. If the resources are not reused by destination B within this time, then the network will automatically recall user A. g. When user A accepts the CCBS recall, then the network will automatically generate a CCBS call to destination B. When the network receives an indication that the destination B is being informed of the CCBS call, the corresponding CCBS request is considered as completed. h. User A always has the possibility to cancel the CCBS service. i. ISUP signaling is required all the way on the complete path (between the calling user A and the called destination B) with the capability of supporting the CCBS call indicator. iv TCAP shall be used for all inter-network bearer-unrelated dialogues used to support the CCBS service.
v The Signalling requirements for implementation of CCBS service shall be as per ITU-T Recommendation Q.733.3.
9.12.16 Call Monitoring -
It allows an authorised supervisor, to listen in on a phone conversation.
9.12.17 Call Parking
The Call Park feature allows a person to put a call on hold at one telephone set and continue the conversation using some other telephone set.
9.12.18 Call Pick-up
The Call Pick-up feature allows a person to answer someone else's call
9.12.19 Call Queuing
Allows multiple calls to be placed in a queue and answered by the next available operator.
9.12.20 Call Recording
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Recording of a phone conversation for later playback. It may be noted that Significant amounts of disk space shall be used (approx. 16KB/sec of recording. Purchase may specify the maximum recording duration for recording the calls.
9.12.21 Call Blocking
This feature lets you identify certain phone numbers, exchanges or area codes from which calls shall be blocked.
9.12.22 Call ID on Call Waiting
Display the phone number of a second caller whilst talking to the first caller.
9.12.23 Distinctive Ringing
Different Rings based on caller ID or selection the caller makes whilst interacting with the auto attendant. For this facility handset must support multiple ring tones.
9.12.24 Music on Hold
The Music on Hold feature allows the system to play pre-recorded music to fill the silence that would be heard by telephone callers that have been placed on hold. Normally, No external device for ‘Music on hold’ is required. While callers are waiting to be connected to an extension, they can listen to music or a recorded message. Choose music from a wide variety of interesting titles, or generate excitement with a promotional message.
9.12.25 Music on Transfer
Music plays when transferring calls between extensions as in case of Music on hold.
9.12.26 Custom greeting
The message heard by callers when the auto-attendant answers incoming calls can be changed to meet one’s specific needs including a professional voice talent recording of the exact greeting purchaser wants callers to hear.
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9.12.27 Personal greetings
Instead of canned messages, individual extension can customize the greeting that one whishes to give to callers. One record his own voicemail greeting, so callers hear his voice and one can say anything one wants.
9.12.28 Call Return / Camping
The Call Return feature allows a called party to place a call back to the calling party of the last received call.
9.12.29 Follow-me / Find-me
The Follow-me feature routes incoming calls to a person trying each number in a pre-configured list until the call is answered.
9.12.30 Message Waiting Indicator (MWI)
The Message Waiting Indicator is an audio or visual signal, the telephone device send to inform that a voicemail message is waiting
9.12.31 Voice mail or VMS
When an incoming call cannot reach the wanted number, it shall be possible to divert the call to an announcement/operator/suitable Voice Mail Service to give the caller appropriate information. Voicemail is a system allowing callers to record and store phone messages and allowing its users to play and distribute stored messages. Called subscriber shall be able to retrieve and manage the recorded messages.
The reasons for the Routing to VMS/announcement shall include:
i. Change of particular extension number including indication of number ii. Dialling of allocated extension number. iii. Dialled extension number is no longer in service. iv. Invocation of' 'Do not disturb' service.
9.12.32 Closed User Group:
i The number of extensions in a closed user group shall be configurable. It shall be possible to add or remove any extension from a group.
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Maximum number of extensions in a user group shall be specified by the purchaser.
ii The number of such closed user groups shall be specified by the purchaser.
iii It shall be possible to administratively assign different facilities/restrictions on different extensions of the same closed user group.
iv It shall be possible for any extension to be member of more than one `closed user group’.
v Intra-group calls shall be possible with short dialling and without dialling the full directory number/extension number.
9.12.33 Parallel Ringing
It shall be possible to allocate any extension as back-up extension to some other extension number. In such cases, it shall be possible to send ringing current/ message on both the extensions in parallel (i.e. simultaneously) and the call shall be routed to the number, which ever called number answers first.
9.12.34 Dial-up Internet Services
There shall be no limitation in accessing and availing any web-based service provided as a ‘dial-up Internet service’.
9.12.35 Fax services
The system shall support group-2 fax services
9.12.36 Automatic Verbal Announcement of Charges applied Service
A user may request an announcement giving the charge for a call, each call of series or the total call charges registered against his name. The request may be made before or after a call. The service will be provided only for DTMF subscribers
9.12.37 Subscriber's Call Charge Meter (Home Meter)
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This facility allows the subscriber to have a meter installed at his premises showing the total call charge units debited
9.12.38 Analogue Telephone Instrument specifications
Analogue type of phone shall have minimum following features:-
i The telephone shall be provided with Display screen which have the following facility:
a Display of called number b Visual indication of incoming call c Duration of successful call d Call waiting indication e Real time clock f Number storage (at it shall be possible to store 50 numbers) g CLI display (up to 16 digits) h Loudspeaker (hands free operation)
ii In addition loudspeaker (hands free operation), Manual pause feature shall also be provided.
iii Dual-tone ringer with volume adjustable in 3 steps (high, medium & low) shall be provided
iv The telephone instrument shall not be damaged when intermittent voltage pulses each of up to 250 ms duration, between earth wire and the line wires connected together, of up to 650 V at 50 Hz are applied.
9.13 Analogue extension
9.13.1 All Analog extensions shall be able to support DTMF telephones. In addition to DTMF telephones analogue extensions shall support Decadic type of telephones
9.13.2 Decadic Dialling
In addition to DTRMF dialing, PABX shall support decadic dialing also. In case of decadic dialling out by analogue extension, decadic pulses shall have the following characteristics:
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i Dial speed shall be 10 ± 0.5 IPS.
ii Make/break ratio shall be 1:2 nominal with break period between 60% to 70%
iii IDP of 800 msec. ± 10%
9.14 ISDN Extension
In addition to user facilities described above in clause 10.11, PABX system shall support the following ISDN bearer services, tele-services and supplementary services on ISDN extensions:
9.14.1 The system shall support the following Number Identification supplementary services: as defined in ITU-T Recommendations I.251, Q.81 and Q951.
9.14.2 Multiple Subscriber Number (MSN):
i As up to 8 terminals can be connected in parallel on the subscriber premises wiring, to call a specific terminal (PC to call a PC, and phone to call a phone) separate number shall be allotted to each terminal. It shall be possible to program each MSN no.
ii The Customer Premise Equipment (CPE) with the MSN supplementary services shall use the least significant “n” digits up to the total number of digits supplied as part of selection process. If a user receives fewer digits than it requires for for terminal selection then the user shall use the available I formation in the called party number information element for its terminal selection.
iii Assigning bearer services, tele-services and supplementary services and out going call barring MSN wise shall be possible.
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9.14.3 Multiple calls: It shall be possible to have two voice, fax or PC "conversations," and one data "conversation" at the same time, through the same ISDN connection.
9.14.4 D-channel messages shall contain signalling data (s-data) as well as packet oriented user data (p-data), provided that the appropriate terminal equipment for packet-oriented data transmission is available where as the Primary Access D-channel shall only be used for signalling information (i.e. for s-data). Packet-oriented user data (p-data) must be separated from the s-data in the PABX and transferred to D-channel. Using the basic channels, transfer of data shall be possible during conversation without disturbing the voice calls.
9.14.5 CLIP: Calling Line Identification Presentation
When an ISDN subscriber receives a call, the calling subscriber number is displayed on ISDN telephone before the called subscriber answers the call. Thus, the called subscriber knows the telephone number of calling party even before answering the call. For example when the subscriber is already in conversation, he may choose to attend the second incoming call depending on the caller’s number displayed. Calling line identification presentation shall be available to all the ISDN subscribers by the default
9.14.6 CLIR: Calling Line Identification Restriction
The calling subscriber with the facility of CLIR will be able to prevent the presentation of his number to the called subscriber (Prevention of CLIP). However, this service will be over ridden by certain agencies such as police and fire services, since they may need to know the identity of the caller in all cases.
9.14.7 COLP: Connected Line Identification Presentation
When the called subscriber answers the call, MSN number of called terminal shall be presented on calling ISDN telephone
9.14.8 COLR: Connected Line Identification Restriction
The calling subscriber with the facility of COLR will be able to prevent the
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presentation of his MSN number to the called subscriber (Prevention of
COLP)
9.14.9 Sub-addressing: It shall be possible to assign separate sub-address to each terminal
9.14.10 Call Offering supplementary services:
The system shall support the following Call offering supplementary services as defined in ITU-T Recommendations I.252, Q.82 and Q952. Coding of Redirecting number information shall be as per ITU-T Recommendation Q.952
9.14.11 CFB: Call forwarding on busy
If the called subscriber is busy, the incoming calls to his number shall be diverted to another number specified by him.
9.14.12 CFNR : Call Forwarding No Reply
If the called subscriber is not available (or does not answer the call), the incoming calls to his number shall be diverted to another number specified by him after a few rings.
9.14.13 CFU: Call Forwarding Unconditional
All the incoming calls to subscriber shall be diverted to another number specified by him. In this case the ring directly goes to diverted number.
9.14.14 Call Completion supplementary services:
The system shall support the following Call completion supplementary services as defined in ITU-T Recommendations I.253, Q.83 and Q953.
9.14.15 Call Waiting (CW)
When any subscriber attempts to make a call to subscriber who is already engaged in another call second incoming call is announced via beeps. The arrival of second call should be displayed as the call waiting message as
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well as is announced via beeps and by pressing the call waiting button. It should be possible to take the second call.
9.14.16 Call Hold (CH)
An external call can be put on Hold on any of the extensions while a second call is to be made or answered. The subscriber can switch between these calls.
9.14.17 Multiple Supplementary Services:
The system shall support the following Multiple supplementary services as defined in ITU-T Recommendations I.254, Q.84 and Q954
9.14.18 3PTY (Three party conference): This enables the user which has one held call and one ongoing call in conversation to joins the held up call also to form a three party conference i.e. all the 3 party are in conversation. It shall be possible to have 3 way conference call with either two external numbers and one internal extension or one external number and two internal extensions.
9.14.19 Additional Information transfer supplementary services
The following Additional Information transfer supplementary services shall be supported as per ITU-T Recommendations I.257, Q.87 & Q.957
9.14.20 UUS: User-to-User Signaling supplementary service 1, 2 & 3 (UUS1, UUS2 & UUS3) using implicit procedure shall be supported. It shall be possible to transfer up to 128 octets of user information in each message.
9.14.21 Mobility and Modification supplementary services:
The following Mobility and Modification supplementary services shall be supported as per ITU-T Recommendations I.258, Q.83 & Q.953.
9.14.22 TP: Terminal Portability
In the subscriber premises up to 8 terminals can be connected to a single ISDN line. These terminals can be in different rooms and also can be on different floors. During conversation it is possible to transfer the call from one terminal to another or even remove the terminal and connect it to
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another socket at a different location. This facility shall be available for calling as well as called subscriber.
9.14.23 CT: Call Transfer
Call can be transferred to any extensions, while first call is in progress.
9.14.24 CD: Call Deflection
It shall be possible to divert the call I ringing phase
9.14.25 CONF: Conference Call
It shall be possible to introduce a common speech path so that all the subscribers in conference are able to speak to each other. Purchaser shall specify the maximum numbers of subscribers in a conference and the maximum numbers of such simultaneous multi-party conferences
9.14.26 AOC–E: Advise of charge at the end of call
It shall be possible send charge information to user side at the end of call. The amount charged for a call, in terms of call units, shall be displayed on the LCD of calling subscriber’s ISDN phone.
9.14.27 AOC–D: Advise of charge during the call
It shall be possible send charge information to user side during conversation. The amount charged for a call, in terms of call units, shall be displayed on the LCD of calling subscriber’s ISDN phone
9.14.28 Speed Calling: It shall be possible to store and dial some frequently used numbers (local or long distance) by dialing only two digits
9.14.29 ECT: Explicit Call Transfer
This will enable the user to connect two calls into one call and shall be able
withdraw from the call
9.14.30 En-block Dialling - PABX shall support en-block dialling from ISDN phone
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9.14.31 Overlap Dialling - PABX shall support overlap dialling from ISDN phone
9.14.32 Intelligent NT – Features
NT shall support the following features:
i. Tone generation: For internal calls, on board tone generator shall provide tones such as RBT, BUSY etc
ii. Emergency Mode: The emergency mode shall provided for the S/T port or one extension of the INT-PBX, once there is local power failure
iii. Status Indicators: LED indicators shall be provided for the Line status and for the extensions also
iv. Operator Mode: It shall be possible to configure analog port 1 as an operator port and all external calls shall land only on that particular operator port. The calls can be then further transferred to the required extensions by the operator.
9.14.33 Inter-working with PSPDN:
i. The system shall support the following facilities for inter-working of ISDN subscribers with X.25 terminals between themselves and also with subscribers of PSPDN
ii. The system shall support transparent access to PSPDN at 64 kbit/s over B- channel as per ITU-T Recommendation X.31in case of inter- working with PSPDN subscribers
iii. Packet mode services in ISDN switch shall be capable of being handled in an external PSPDN connected to ISDN switch on ETSI-PHI 300 099 (1992) interface. PABX shall support the required software and hardware, if any, to enable this inter-working.
9.14.34 ISDN Telephone Instrument:
The ISDN Phones should have at least the following features:
i Tilt able Display 40 alphanumeric characters display
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ii Volume control Keys-Increase and Decrease iii Programmable ringer, loudness and tone character iv Feature keys. - Inclusive of Mute, Hands free and Headset function keys, Enquiry key, Transfer key etc. v Message waiting indicator LED vi Fully duplex vii Menu driven function for feature access viii Menu Navigating Keys ix Min 12 Programmable Keys x Adapter position for ISDN/ Analog Phone/ digital phone Interface xi Support for Master slave configuration xii Name dialling xiii Alpha numeric keypads xiv Automatic Call back xv The subscriber unit should be powered from the system itself (over the same subscriber line pair).
xvi The display must show at least the following information: a Time and date b Own telephone number c Telephone status d Incoming/outgoing call numbers i.e. caller name/number display e Duration of call in case of outgoing successful call
9.15 Wireless Extensions:
In addition to user facilities described above, wireless extensions shall
support the following facilities/functions:
9.15.1. CLIP: Calling Line Identification Presentation
When a DECT subscriber receives a call, the calling subscriber number should be displayed on his telephone before the called subscriber answers
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the call. Thus, the called subscriber knows the telephone number of calling party even before answering the call. For example when the subscriber is already in conversation, he may choose to attend the second incoming call depending on the caller’s number displayed. Calling line identification presentation shall be available to all the DECT subscribers by the default.
9.15.2. Base Station
i The details regarding the nominal Transmitter (TX) output power of base station attached to the PABX and the range of adjustment possible shall be furnished by the equipment supplier. It shall not exceed the maximum limit specified in the standards followed for the air-interface.
ii The base station of the PABX shall be, as far as possible, located at a central place of the building/premises of the user. Locating the base station out side the building/premises shall not be permitted iii Spurious Emission: The spurious emission at Antenna port of the equipment shall conform to either ITU-R 329-6 or the value specified in the standards followed for air-interface 9.16 IP Extensions/Multi-Media PC
In addition to user facilities described above, PABX shall support the follwing additional features on IP extensions:
9.16.1 Conference Bridging: It shall be possible to create a conference call between multiple parties at multiple locations using different phone types; e.g. conference call a local extension, remote fixed line, mobile and VoIP connection all in one conference. The conference administrator can mute, activate and kick participants. Outside participants (clients, suppliers etc…) can join the conference in a variety of ways including dialing into the conference extension from the digital receptionist, being transferred in or by entering a conference code that can be prompted by the digital receptionist. Outside participants are only limited by line availability
9.16.2 Dial by Name: It shall be possible to dial a subscribbers by their name using the phones numeric keypad (using the same technique used when sending an SMS)
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9.16.3 Local and Remote Call Agents: It shall be possible for user to log onto the PABX from any phone using a login ID allowing them to make and take calls as if they were in their office.
9.16.4 Automated directory: Callers shall be given access to a company directory when they don't know a specific extension number. Options of Dial-by-name directory on either the first name or the last name of the employee shall be available.
9.16.5 Call screening: The PABX administrator shall have the facility to enable or disable call screening for the entire company. If it is enabled, each employee can decide whether to turn screening on or off.
When call screening is on, callers are asked to state their names before the call is transferred to the extension owner. This way the extension owner can know exactly who the caller is before picking up the call.
9.16.6 Easy, web-based extension configuration: While overall system configuration shall be done by a system administrator, many functions and features are controlled directly by each extension owner. This gives flexibility to the employee, while reducing the load on the administrator. Typical configuration activities by extension owners include establishing follow-me lists, setting voicemail and fax forward options, checking voice and fax messages, and logging into ACD queues etc. The web configuration tool should be straightforward and easy to use. It shall be possible to do most subscriber functions from any touch-tone telephone
9.16.7 Call-answer security: With follow-me calling, one can send his calls to any phone. Some of these phones might be used by other people. Call-answer security lets user set a password that must be entered before a call is actually connected. If the password is not given, the call will move to the next number in the follow-me list, or to voicemail
9.16.8 Never full: Systems shall have dynamic voicemail storage. System shall automatically open more message space for the user when his space gets full, moving older messages out to make room for newer messages. The
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older messages are kept for pre-specified period, giving you ample time to retrieve them.
9.16.9 Centralized messages: Some user may have several phones, each with its own voicemail. It shall be possible to centralize all the messages in one place. User don't have to search through multiple systems, he/she just look in one place.
9.16.10 Notifications: It shall be possible to notify the user receipt of any voice message or fax for him. Notification can be email notification, pager signals, web alerts and text messaging on his/her cell phone.
9.16.11 Fax viewing options: PABX shall handles incoming faxes automatically, without the need for a fax machine. The faxes shall be delivered as .pdf files, and can be viewed through web tool or can be forwarded to user email for viewing.
9.16.12 Direct Inward System Access (DISA): The Direct Inward System Access feature allows a calling party to access internal features of the system from an outside telephone line.
9.16.13 Video communication: With respective cameras installed, a terminal user can directly originate a video call to the opposite party and the appropriate video quality can be determined depending on the network bandwidth.
9.16.14 File transfer: Transfers files to another user or department where the received files can be saved under a particular directory or a specified directory.
9.16.15 Application share: A terminal user shares an application; then the opposite party can use this shared application remotely.
9.16.16 Electronic whiteboard: Both parties can write and draw on the same picture, for example, for discussion purposes. This is applicable to many occasions such as remote teaching and technical exchange.
9.16.17 Content release: The contents of advertisements and media streams can be released through a multimedia terminal, and a platform can be used to selectively locate the user or to immediately release.
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9.16.18 Instant messaging: Allows real-time communication by means of text between one terminal user and another who has already logged in
9.16.19 Customized Phone Profile: It shall supports web based customizable skin templates. Such functionality allows user to update the underlying image files and create customized softphones. Additionally, each provider can create multiple phone profiles with different design, language, and functionality and use them to target different customer groups.
9.16.20 webFones PBX Features: Following webFones features shall be supported:
i Multi-Location Functionality: User shall be allowed full access to all the features of the system from any user terminal.
ii Voicemail via Email: PABX sends digital voicemail sound files and Caller ID via email. It shall be possible to scan, forward and organize voicemail messages. This shall be in addition to traditional voicemail retrieval (dialing a voicemail box).
iii Call Record: It shall be possible to make MP3 recordings of phone conversations, simply by pressing the "Record" button on IP phone. Recordings shall be sent automatically o user’s email box after the call ends.
9.16.21 Selective Call Forward
9.16.22 Selective Call Rejection
9.16.23 Instant messaging (IM) and presence:
Presence information and notification shall be provided as below:
i Information that client is registered ii Information that Client is currently engaged in an Instant Messaging Session. iii SIP clients shall communicate with SIP Server using SIMPLE protocol. iv Click-to-Call - Initiate a call from any directory entry
9.16.24 It shall be possible to implement and provision services, based on following capability types, for which APIs shall be provided:
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– Identification – Authentication – Authorization – Location – Presence and Availability – Group management (CUG/VPN) – Call Control: • Generic • Multi-Party • Multimedia • Conference – Session Control – Service Subscription (Registration) Management – User and Terminal Profile Management – Generic messaging – Multimedia messaging – Push – Content based charging – Emergency Communication Management – Address list management – Prioritized communication /traffic handling – Service Independence - service capabilities which will allow IP multimedia applications to be deployed in a vender independent manner. – Application Service inter working - Allows inter working of application service protocols. This capability should allow inter working between application services and network entities for execution of the service
9.16.25 It should provide the capabilities to:
Determine the location of the target end point.
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Determine the media capabilities of the target end point via Session Description Protocol (SDP). Determine the availability of the target end point. Establish a session between the originating and target end point if the call can be completed. Handle the transfer and termination of calls. SIP supports the transfer of calls from one end point to another
9.17 Suggestive specifications of IP hard phone:
i The IP Hard phone should be SIP enabled ii Keypad, programmable function key with LED indicator, dialog keys and Plus/ Minus Keys iii 40-digit alphanumeric display with background illumination iv Intelligent dialog key control for function selection in the display by scrolling in the menu and selecting v Open listening vi hands free talking (full duplex) vii integrated headset-set port viii power over LAN ix support of wide band codec x support of DHCP, SNMP , FTP xi update of features using FTP xii 10Base T/100BaseT Ethernet interface with mini-switch (VLAN) xiii support of CTI (TAPI) xiv plug-in adapters can be used xv Codec: G.711/723/729/722 xvi Power on LAN toIEEE802.3af xvii Silence suppression (VAD)
9.17.1 Suggested Video Resolution:
System shall support minimum video resolution of
a Pro-Motion interlaced video (60/50 fields full-screen video for NTSC/PAL) b 4SIF (704 x 480)
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c 4CIF (704 x 576) d SIF (352 x 240) e CIF (352 x 288)
9.17.2 PABX shall comply the following Protocols/ specifications for IP terminals:
'SIP: Session Initiation Protocol', RFC 3261 Session Description Protocol, RFC 2327 An Offer/Answer Model with the Session Description Protocol (SDP), RFC 3264 Reliability of Provisional Responses in the Session Initiation Protocol (SIP), RFC 3262
9.17.3 Following RFC shall be supported: 2246, 2543, 2782, 2806, 3263, 3265, 3266, 3311, 3312, 3323, 3326, 3420, 3428, 3455, 3515, 3556, 3588, 3665, 3725, 3856, 3863, 3880, 3891, 3911, 3966, 4028
9.17.4 IP Phones must support 802.1x (EAP-MD5/AES or better) for authentication and access control to the network i.e. user shall be connected to network only after he has passed the authentication process
9.17.5 The IP Phone must support built in Ethernet switch to cascade PC with the phone so that single I/O port is used to connect both IP phone and PC
9.17.6 Ethernet Interface:
9.17.6.1 QoS management
i. The interface shall support QoS marking and mapping as well as Priority marking and mapping
ii. The interface shall support re-map of ToS bits and DiffServ code points between networks for QoS enforcement.
iii. The interface shall support policy based admission control features on the basis of call details, QoS and bandwidth.
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iv. The interface shall support flow reporting which includes functions such as Quality and Service Level Agreement (SLA) monitoring.
v. The QoS parameters as below, shall be supported
a) ‘R’ value greater than equal to 80
b) Jitter < 5ms
c) Packet loss < 0.1%
vi. Codec: The interface shall support the procedures for Codec negotiation, in association with the control equipment. It shall support G.729A, G.729B, G.729AB, G.725A, AMR, H.264 (for video) and T.38 codec as per ITU-T recommendations.
9.17.6.2 Traffic Measurement and Recording
i. Traffic monitoring and shaping feature shall be provided. Traffic report shall be generated for I/C and O/G traffic.
ii. The following measurements shall be provided:
a) Processors occupancy.
b) Total traffic handled in Mbps.
c) Total traffic carried by the Ethernet Interface. The total RTP traffic carried by the Ethernet Interface. Rate, ratio and total number of error packets.
d) Number of Ethernet frames sent out. .
e) Number of re-transmitted control/signalling messages during a period.
f) RTCP traffic handled.
g) Traffic handled per codec type.
iii. It shall be possible to activate and record/print the measurement of delay, packet loss per session also.
iv. The above measurements, shall not unduly affect the call handling capacity of the processor.
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9.17.6.3 The system should support Wireless IP hard phones. These phones get controlled by the PBX for telephony features and gets physical connectivity using 802.11a/b/g interface from underlying WiFi network
9.17.6.4 The PBX also should support Soft IP phone that emulates all the functionalities of the hard IP Phone and get installed on the user’s PC/Computer
9.17.6.5 The IP PBX should have native SIP-CTI (TR-87 standard) connectivity for integration with computer applications.
9.18 Other Extension Requirements: Only applicable extensions requirements may be prescribed by the purchaser
9.18.1 If different types of extensions are required i.e. IP terminals as well as PSTN terminals then PSTN/ ISDN/ Wireless (DECT) subscribers shall be able to make calls to IP terminals using PSTN numbering scheme. IP terminals shall be able to access PSTN services using PSTN numbering scheme i.e. by their E.164 address.
9.18.2 Inter-working for all types of extensions (required) shall be possible from each type of terminal (extension) to all other types of terminals (Analog/ ISDN/ IP Phone/ Wireless/ Multimedia PC etc.). The IP PABX shall provide address translation necessary for inter-working of services, in all cases. In case, it is not possible to complete the call or the service due to any limitation of a particular type of terminal, the calling and/ or called subscriber (as applicable) shall be informed through audio/ video messaging/tones.
9.18.3 Analogue Subscriber Line (Applicable for analogue extensions only)
9.18.3.1. The system has to feed a minimum of 30 mA of transmitter current at a nominal battery voltage of PABX.
9.18.3.2. The maximum open circuit D.C. voltage presented between the ‘a’ and ‘b’ wires of the NTP of an analog interface should not exceed 78 V as per ETSI TR 101 188.
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9.18.3.3. The maximum loop current which the PABX can feed into the subscriber loop shall not exceed 60 mA
9.18.4. ISDN Extension (Applicable for ISDN extensions only)
9.18.4.1. The system shall support Basic rate access at 144 kbit/s i.e. two basic channels (2B) of 64 Kbps and one data channel (D) of 16 kbps, for providing access to the extension subscribers. Bearer channels (or B channels) transmit user information at relatively high speeds, while separate Data channels (or D channels) carry call set-up, signaling and other information.
9.18.4.2. These three channels (2B + D) are transmitted using only two metallic wires by converting two wires into virtual 4 wire by wiring hybrid and switching the data in two directions of transmission.
9.18.4.3. Multiple devices on the same line. The telephone line shall be terminated on NT (Network Termination) on which it shall be possible to connect up to eight separate ISDN terminals. These ISDN terminals can be Digital Phone, Video Telephones, G4 Fax machines, PC based terminals, Video conferencing equipment which can be linked to a single ISDN connection. In addition existing terminals such as Analog phone, G3 Fax can also be connected using suitable adopters. Also one terminal can be shifted by plugging out from one digital socket to other.
9.18.4.4. ISDN customer premises equipment used on the extension lines shall be interface approved as per TEC IR/CPE-02.
9.18.5. Wireless Extensions
9.18.5.1. Introduction: DECT or Digital Enhanced Cordless Telecommunications is an ETSI standard for digital portable phones. The DECT standard fully specifies a means for a portable unit to access a fixed telecoms network via radio. Connectivity to the fixed network is done through a base station or "Radio Fixed Part" to terminate the radio link
9.18.5.2. Connectivity with base station: The Base Station shall be connected to PABX by a 2 wire twisted pair cable. The connection to the Base Station should be independent of polarity.
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9.18.5.3. Overlap: Overlap of the radio coverage of adjacent base stations must allow sufficient time for the user to move from one area to another and the base station to hand the call over. If the overlap is too small (or the user is moving faster) then the call may be disconnected during the handover. An overlap of 10-15 meters is recommended.
9.18.5.4. Shadows: A shadow is an area where there is no radio coverage. There should be minimum Shadow area.
9.18.5.5. The PABX with wireless extensions shall be based on the state-of-the-art digital technology using low power in the frequency band of 1880-1900 MHz or 864 - 868 MHz or any spare frequency band allotted by Wireless and Planning & Co-ordination (WPC) wing of DoT. For operation of the PABX with wireless extensions at a specific location, necessary permission/license from Service Provider and frequency clearance from Wireless and Planning & Co-ordination (WPC) wing shall be obtained. WPC will give clearance for use of these frequencies on case-to-case basis.
9.18.5.6. The PABX with wireless extensions may also have the capability to work in conjunction with an existing PABX when located in the same premises.
9.18.5.7. It shall be possible to connect multiple handsets to one base station and one phone line socket.
9.18.5.8. The system shall provide interference-free wireless operation in the coverage area. It shall be immune to interference from Wi-Fi or video senders
9.18.5.9. Wireless extensions: Fixed instruments or hand-held sets can be used for extensions. The maximum operating range of the wireless extensions will be restricted to private premises only or 500 meters from the PABX whichever is less. Details regarding the coverage area for mobility of the equipment offered shall be furnished by the equipment supplier.
9.18.5.10. Wireless terminals used for extensions shall conform to relevant international/ national standards used for the air-interface. A test report and certificate from a recognized accredited international/ national test agency shall be submitted by the equipment supplier.
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9.18.5.11. The Grade-Of-Service (GoS) offered to the wireless extension subscribers and the number of wireless extensions provisioned, shall be indicated by the equipment supplier.
9.18.5.12. It shall be possible to connect multiple handsets to one base station and one phone line socket
9.18.5.13. Each hand set shall be allotted a unique IPEI (International Portable Equipment Identity) which shall be authenticated for each call.
9.18.5.14. When no traffic channel on the air interface is available due to busy or faulty condition of different channels, busy tone or a suitable announcement shall be given to calling subscriber.
9.18.6. IP Extensions/Multi-Media PC
9.18.6.1 General: IP extensions/ MM PC can be connected to PABX only in case PABX is based on IP technology.
9.18.6.2 PABX shall provide interface for connecting IP Telephones (SIP terminal or H.323 terminal) or PC based Video Phone with suitable Client software” which shall work as extensions.
9.18.6.3 It shall be possible to use IP phones of any make, interface approved as per TEC IR NO. IR/IPP-01.
9.18.6.4 IP Terminal terminals shall be connected to PABX over Ethernet as per IEEE 802.3
9.18.6.5 All IP extensions shall support IP addressing as per version 4 (IPv4) or IP version 6 (Ipv6). The same shall be indicated in the certificate.
9.19 Subscriber Line Testing:
9.19.1 Analogue extension line testing-
i It shall be possible to carry out Line tests from either subscriber’s premises or from the PABX system. ii The system shall provide all the tones and frequencies required for the Line tests.
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iii It shall be possible to automatically test a subscriber line and group of lines using Man-Machine Commands from O&M terminal. The results shall be displayed and logged into a file for retrieval whenever required.
iv The system shall provide the following measurements/adjustments from the subscribers premises Measurement of line insulation. Bell adjustment. Measurement of microphone current. Test of dial speed. Test of DTMF key pad
v The system shall provide the following measurements from using GUI based terminal Interference voltages on the two line wires; 50 Hz AC voltage, DC voltage. Insulation resistances of line wires with respect to ground or between ‘a’ wire and ‘b’ wire. Line capacitance (including the instrument capacitor), in order to check the line continuity.
vi Loop resistance of the line. The test procedures shall facilitate isolation of faults between ‘line side’ and equipment side.
9.19.2 ISDN line testing
It shall be possible to conduct the following tests on an individual ISDN BRA subscriber line and group of BRA lines using Man-Machine Commands. The results shall be displayed on the terminal and shall be logged into a file for retrieval whenever required.
i BER test ii Line insulation test iii Line capacitance measurement iv Continuity test
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9.19.3 Wireless Line testing:
It shall be possible to conduct the following tests on an individual subscriber line and group of subscriber lines (up to base station) using Man-Machine Commands. The results shall be displayed on the terminal and shall be logged into a file for retrieval whenever required. i. Interference voltages on the two line wires ii. Line insulation test iii. Insulation resistances of line wires with respect to ground or between ‘a’ wire and ‘b’ wire iv. Loop resistance of the line. v. The test procedures shall facilitate isolation of faults between ‘line side’ and equipment side. vi. It shall be possible to measure the transmitted frequency and it’s level 9.20 Signalling between PABX and extension:
9.20.1 Analogue extension (Applicable only if approval for analogue extensions is sought)
9.20.1.1 PABX shall use DTMF signaling between the subscriber line and PABX as per ITU-T recommendation Q.23
The number of digits/characters used for signaling shall be 12 and a set of two frequencies shall be used for each digit/character. Following two sets of frequencies shall be used:
Low frequency group : 697,770,852,941 Hz
High frequency group : 1209, 1336, 1477 Hz
Digit Frequencies Digit Frequencies
used used
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Digit 1 697 & 1209 digit 7 852 & 1209
digit 2 697 & 1336 digit 8 852 & 1336
digit 3 697 & 1477 digit 9 852 & 1477
digit 4 770 & 1209 digit 0 941& 1336
digit 5 770 &1336 * 941 & 1209
digit 6 770 & 1477 # 941 & 1477
9.20.1.2 Each transmitted signaling shall be within ± 1.8% of the nominal frequency.
9.20.1.3 The total distortion products (resulting from harmonic or inter-modulation) must be at least 20 dB below normal frequencies.
9.20.1.4 Level difference between any two frequencies forming a signal should not be more than 2 dB and level of the higher frequency component of the signal shall be of higher level.
9.20.1.5 During DTMF signaling, the d.c. loop shall be maintained.
9.20.1.6 In case of DTMF redial, the on/off period shall be at least 40/40 ms.
9.20.1.7 The level of the signals shall be as follows:
Low group -8 ± 2 dBm
High-group -6 ± 2 dBm
Pre-emphasis 2 ± 1 dB
9.20.1.8 It shall be possible to connect any telephone instrument of any make which is approved TEC approved as per TEC GR No. GR/TEL-02 without effecting the system functioning
9.20.1.9 Calling Line Identification Presentation (CLIP)
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System shall support DTMF signalling for CLIP. i. It shall be possible to send the CLIP on any extension. In case of internal call, extension number shall be displayed on the called extension. In case of call received from public network, CLIP received from the network shall be send to called number without any change. ii. In case of calls originated from the PABX to public network, PABX shall send CLI information which shall include “Access code of PABX” and the extension number. iii. In case of Call forwarding by 'B' and when call is made from 'A' to 'B', then 'C' shall get CLI of 'A'. iv. The PABX shall be able to send up to 16 digits of CLI information v. Calling & called extensions may be of any type, it shall be possible to send the CLIP. vi. The duration of DTMF tone on line shall be 50 ms ± 2 ms. vii. The inter-digital pause i.e. the duration between consecutive digits shall be 50 ± 2 ms. viii. If the call is answered during the transfer of CLI information by the system, the call is regarded as being answered and the system shall abandon further transmission of CLI and put through the call. ix. The sequence of CLI digits sent from exchange will be in the same order as normally dialled by subscribers, i.e. area code followed by subscriber number. x. PABX shall not support “Calling Line Identification Restriction” (CLIR) feature. This feature restricts the presentation of the calling party’s number to the called party.
9.20.2 ISDN Extensions (Applicable only if the approval for ISDN extensions is sought)
Integrated services Digital Network (ISDN) provides different services like voice,
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data and image by means of existing network. An ISDN subscriber can establish two independent calls simultaneously on the existing path of telephone line. These simultaneous calls can be of any type, speech, data or image. Typical connection of various ISDN customer premises equipment’s is given in Fig. 2 (It is only is typical connection diagram but it is not mandatory)
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ISDN PHONE SOCKET P RS TERMIAL C 232 ADAPTOR
ANALOG PHONE
G III FAX
PC with ISDN ADD- EPAB U S NETWORK ON-CARD X TERMINATION BRI DESKTOP VEDIO CONFERENCING NT1 (work at 128kbps)
VIDEO PHONE
Fig. 2 Typical Connection of various ISDN Customer Premises Equipment
9.20.2.1. The PABX shall support the layer 1, 2 and 3 of the ISDN user network interface conforming to TEC generic requirement S/ISN-01.
9.20.3 Wireless extensions (Applicable only if the approval for wireless extensions is sought)
9.20.3.1 For connection between the PABX and wireless extensions, coding shall conform to ETSI DECT standards ETS 300 175 and 300 176 PABX.
9.20.3.2 The system shall provides encryption with the DECT Standard Cipher (DSC) for data security
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9.20.3.3 For interoperatability between various DECT equipmet from different manufacturers for basic fuctions, system shall coform to GAP profile defined within the DECT standards ETS 300 175 and 300 176.
9.20.4 IP Extensions (Applicable only if the approval for IP extensions is sought)
9.20.4.1 IP terminals shall be connected over Ethernet interface as per IEEE802.3
9.20.4.2 Signalling between IP extension or Multimedia PC and IP-PABX shall be provided either
using application-layer control (signalling) protocol, Session Initiation Protocol (SIP) as per IETF RFC 2543 and RFC 3261 for various multimedia sessions.
or
H.323 as per ITU-T H.323 protocol
9.20.4.3 If H.323 protocol is used for signalling between IP terminal & PABX, it shall be as per ITU-T H.323 protocol. If SIP protocol is used for Signalling between IP terminal and PABX, it shall be as per TEC National Standards for SIP No. SD/ SIP- 01
9.21 VMS System: The offered system shall have integrated server/card based ‘Voice mail system’, with dynamic storage of each mailbox. Users should be able to access the system internally or remotely from any phone & should be able to record standard/personal messages within the mailbox. The system should be able to inform the outside caller about the exact status of the desired extension (no answer/ busy). The System should also support recording of name and personalized greeting within each mailbox. System administration should be user friendly and the system should be windows based. The application and development tool shall allow call flow transaction to be written & modified at site using simple menu driven command. The on line updating, changes/ modification in application should be dynamically loaded/ assigned without switching off or disturbing the services. Call answering or call/ record message should be with time stamp. Purchaser shall specify the following parameters of VMS:
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Minimum Number of Mailboxes Minimum Recording time Number of Simultaneous accesses ports
9.22 In addition to above desribale requirements, following parameters shall also be specified by the purchaser as per his/her requirements
9.22.1 Capacities of the PABX
Following parameters required for dimensioning purpose may be specified by the
tendering authority:
i. Number of extensions of each type ii. Type of interface(s) towards public network and number of interfaces of each type. Communication Protocols may also be specified. iii. Call Holding time per subscriber and operator. iv. % Mix of Intra-PABX calls and calls to public network v. Optional functions and user applications, if required.
9.22.2 Various dimensioning parameters e.g. traffic handling capacity, storage type and volume, synchronous or asynchronous etc. shall be indicated.
9.22.3 Exact requirement for number of total and concurrent users shall be indicated.
9.22.4 In case of IP-PABX, type of Ethernet backbone (e.g. 10 Mbps—10Base-T Ethernet, 100 Mbps—Fast Ethernet or 1000 Mbps) shall be indicated.
9.22.5 If any additional Codec support is required, same may be specified.
9.22.6 Details of hardware, software and number of operator terminals for operation, administration and maintenance may be specified.
9.22.7 Tools and Testers required, if any
9.22.8 The period for which the maintenance spares are required, may be specified by the tendering authority.
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9.22.9 Battery Requirements. Power requirements of AC and DC power supplies. Number of sets to be supplied
9.22.10 Documentation: Number of copies (hard/soft) to be supplied.
9.22.11 Qualitative Requirements (QR): The purchase shall specify quality standards like ISO 9002 or ISO 9001: 2000 certification.
9.22.12 Environment Conditions: The purchaser shall specify the requirements of Environment Conditions that the system shall satisfy as specified in Quality Measure Manual for relevant category of equipment.
9.23 PABX may support the deployment of network separation methods such as SBC, firewalls, virtual routers (sub-netting) at the edge of service provider’s network. (Applicable only in case of Type-7 interface)
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10.0 Specific items to be mentioned in the certificate 1. Type(s) of interface tested. 2. a) Type(s) of Extension(s) used for testing (e.g. Analogue Phone, ISDN Phone, IP phone etc.) was/were used for testing.) b) In case of IP Terminal, (IPv4/IPv6) should be indicated. 3. Type of Physical Interface (In case of IP interface only) (Ethernet type 10/100/1000 Mbps /twisted pair/coaxial/ fiber etc) a) Type of IP signalling (SIP /H .323), (In case of IP interface only) b) Signalling supported i.e. DTMF & Decadic, whichever applicable (In case of Type-1 interface) 4. Additional codec supported, if any. 5. Software required to read CDRs – Inbuilt/supplied separately 6. This Certificate does not cover the functions, features, performance, capacity etc. of the equipment other than those necessary for meeting the interface requirements. 7. Approval of PABX against this IR, does not entitle the user to connect the equipment to the network of Internet Service Provider’s (ISP).
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ABBREVIATIONS
ADPCM Adaptive Differential Pulse Code Modulation BRI Basic Rate Interface BICC Bearer-Independent Call Control CCBS Call Completion on Busy Subscriber CDR Call Detail Record CLI Calling Line Identification CPE Customer Premise Equipment CSCF Call Session Control Function CSH Called Subscriber Held Db Decibels dBm The absolute power level in Decibels DDF Digital Distribution Frame DECT Digital Enhanced Cordless Telecommunication DEL Direct Exchange Line Direct Inward Dialling DID Department of Telecommunications DoT DTMF Dual Tone Multi-Frequency EMC Electro-Magnetic Compatibility GoS Grade of Service GR Generic Requirements Hz Hertz IAC Interface Approval Certificate I/C Incoming IEC International Electrotechnical Commission IEEE Institute of Electrical and Electronics Engineers IETF The Internet Engineering Task Force IR Interface Requirements IDP Inter Digital Pause IP Internet Protocol
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IPS Impulses Per Second IR Interface Requirements ISDN Integrated Service Digital Network ISP Internet service provider ITU International Telecommunication Union IVRS Interactive Voice Recognisation System LCL Longitudinal Conversion Loss MFC Multi-Frequency Compelled MVI Message Waiting Indication MM-PC Multi media PC MSN Multiple Subscriber Number O/G Outgoing PABX Private Automatic Branch Exchange POTS Plain old telephone system PCM Pulse Code Modulation PRI Primary Rate Interface PSTN Public Switched Telephone Network RFC Request For Comment SBC Session Border Controller SIP Session Initiation Protocol TDM Time Division Multiplexing TEC Telecommunication Engineering Centre TX Transmitter VMS Voice Mail Service VoIP Voice Over IP WPC Wireless Planning & Coordination ======End of the document ======
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