Worldwide, a large number of people interact with each other by means of online chatting. There has been a significant rise in the number of platforms, both social and professional, such as WhatsApp, Facebook, and Twitter, which allow people to share their experiences, views and knowledge with others. Sadly enough, with online communication getting embedded into our daily communication, incivility and misbehaviour has taken on many nuances from professional misbehaviour to professional decay. Generally flaming starts with the exchange of rude messages and comments, which in turn triggers to higher scale of flaming. To prevent online communication from getting downgraded, it is essential to keep away the hostile users from communication platforms. This paper presents a Security Detection Model and a tool which checks and prevents online flaming. It detects the presence of flaming while chatting or posting blogs, and censors swear words as well as blocks the users from flaming.
A private voice network can save companies money on telephone bills and ensure callers can reach the desired extension on the first call. Designing such a network involves choosing between direct inward dialing (DID), private leased circuits, public switched telephone network (PSTN) overflow, or virtual private networks (VPN). The optimal solution depends on factors like call volumes, number of sites, and whether a wide area data network exists that could incorporate voice services. Numbering plans are also important to integrate different sites on a private network or VPN.
Performance evaluation of bandwidth optimization algorithm (boa) in atm networkEditor Jacotech
domains: none of them are suitable, alone, for the wide range of traffic services expected in ATM-based networks. Therefore, some integration of these basic schemes should be considered. In this paper, we propose a new traffic control algorithm, called the Bandwidth optimization Algorithm (BOA). BOA is a multi-level control algorithm that attempts to optimally manage network resources and perform traffic control among a wide range of traffic services in ATM-based networks. The basic objective of BOA is to meet the quality of service requirements for different traffic sources, while making the best possible use of network bandwidth. In addition. BOA attempts to minimize network congestion in a preventive way.
Operators interconnect their networks to allow customers to communicate across networks. Interconnect agreements define pricing, quality of service, billing processes, and more. Interconnect billing involves calculating payments between operators for incoming and outgoing call traffic based on call detail records. Settlement and reconciliation processes are used to exchange CDRs and invoices and resolve any discrepancies between operators.
Deploying a cloud-based contact center with an existing premises-based phone ...InteractiveNEC
Do you still have a premises-based phone system but looking to move your contact center into the cloud? With UNIVERGE BLUE ENGAGE Contact Center (CCaaS), you can deploy your cloud-based contact center with your existing on-premises phone system. Learn more about UNIVERGE BLUE ENGAGE today on our website https://www.univergeblue.com/cloud-services/engage.
Simple and clearly ,Brief defined Terms which i learnt during My internship in PTCL.
Further more if you have any problem you can E.mail me.
hopefully will help you guys
The document is an internship report submitted by M. Umar Khalid to the University College of Engineering & Technology at the Islamia University of Bahawalpur, Pakistan. The report details the internship conducted at Pakistan Telecommunication Company Limited (PTCL). It provides an overview of PTCL's organizational structure, services offered such as landlines, broadband and wireless, and technical aspects of the network including the main distribution frame (MDF) and digital subscriber line (DSL) technology. It also describes the next generation network (NGN) architecture and equipment used by PTCL such as the Alcatel exchange switch.
Mobile number portability allows mobile subscribers to retain their existing phone numbers when switching service providers. It was implemented in India in 2011 to promote competition between providers. Before MNP, subscribers were reluctant to switch providers as it meant changing their phone number. MNP uses a centralized database and all call query routing to route calls to ported numbers. It benefits users through choice, benefits providers through competition, and benefits the government through increased revenue and investment. However, MNP only applies to mobile numbers within service areas and has a 90 day waiting period between provider switches.
A private voice network can save companies money on telephone bills and ensure callers can reach the desired extension on the first call. Designing such a network involves choosing between direct inward dialing (DID), private leased circuits, public switched telephone network (PSTN) overflow, or virtual private networks (VPN). The optimal solution depends on factors like call volumes, number of sites, and whether a wide area data network exists that could incorporate voice services. Numbering plans are also important to integrate different sites on a private network or VPN.
Performance evaluation of bandwidth optimization algorithm (boa) in atm networkEditor Jacotech
domains: none of them are suitable, alone, for the wide range of traffic services expected in ATM-based networks. Therefore, some integration of these basic schemes should be considered. In this paper, we propose a new traffic control algorithm, called the Bandwidth optimization Algorithm (BOA). BOA is a multi-level control algorithm that attempts to optimally manage network resources and perform traffic control among a wide range of traffic services in ATM-based networks. The basic objective of BOA is to meet the quality of service requirements for different traffic sources, while making the best possible use of network bandwidth. In addition. BOA attempts to minimize network congestion in a preventive way.
Operators interconnect their networks to allow customers to communicate across networks. Interconnect agreements define pricing, quality of service, billing processes, and more. Interconnect billing involves calculating payments between operators for incoming and outgoing call traffic based on call detail records. Settlement and reconciliation processes are used to exchange CDRs and invoices and resolve any discrepancies between operators.
Deploying a cloud-based contact center with an existing premises-based phone ...InteractiveNEC
Do you still have a premises-based phone system but looking to move your contact center into the cloud? With UNIVERGE BLUE ENGAGE Contact Center (CCaaS), you can deploy your cloud-based contact center with your existing on-premises phone system. Learn more about UNIVERGE BLUE ENGAGE today on our website https://www.univergeblue.com/cloud-services/engage.
Simple and clearly ,Brief defined Terms which i learnt during My internship in PTCL.
Further more if you have any problem you can E.mail me.
hopefully will help you guys
The document is an internship report submitted by M. Umar Khalid to the University College of Engineering & Technology at the Islamia University of Bahawalpur, Pakistan. The report details the internship conducted at Pakistan Telecommunication Company Limited (PTCL). It provides an overview of PTCL's organizational structure, services offered such as landlines, broadband and wireless, and technical aspects of the network including the main distribution frame (MDF) and digital subscriber line (DSL) technology. It also describes the next generation network (NGN) architecture and equipment used by PTCL such as the Alcatel exchange switch.
Mobile number portability allows mobile subscribers to retain their existing phone numbers when switching service providers. It was implemented in India in 2011 to promote competition between providers. Before MNP, subscribers were reluctant to switch providers as it meant changing their phone number. MNP uses a centralized database and all call query routing to route calls to ported numbers. It benefits users through choice, benefits providers through competition, and benefits the government through increased revenue and investment. However, MNP only applies to mobile numbers within service areas and has a 90 day waiting period between provider switches.
Mobile Number Portability completed its 8 months, investment of approximate Rs.10,000 Crore, & according to data from the telecommunications regulator, about 13 million subscribers changed their service providers until the end of June
Mobile number portability with IEEE MaterialSachin Saini
This presentation file contains all the basic information about Mobile Number Portability, its implementation, routing mechanism and its benefits to user and telecom operator.
Dear students get fully solved assignments
Send your semester & Specialization name to our mail id :
“ help.mbaassignments@gmail.com ”
or
Call us at : 08263069601
(Prefer mailing. Call in emergency )
MNP, or mobile number portability, allows cellular users to change their mobile service provider while keeping their original phone number. It involves a port out process to deactivate a subscriber from their existing provider and a port in process to activate them on the new recipient provider's network. The porting transaction is facilitated through a number portability clearing house and gateways to intercept and route calls to ported numbers on the new provider's network.
Estimating market share through mobile traffic analysis linkedinAsoka Korale
This document discusses methods for estimating market share based on traffic analysis. It proposes expressing each operator's traffic contribution in terms of a known operator's intra-network traffic volume. By doing so, the estimate is not dependent on the composition of subscriber bases or tariffs. An example is provided to illustrate how outgoing traffic from one operator to others can indicate their relative market shares. Adjustments are described to account for differing on-net and off-net tariff rates. Validation through call-by-call analysis is suggested. The approach provides objective and easily automated estimates of market share proportions.
This document defines terminology used in contact centers. It provides definitions for over 50 terms including abandoned call, average handle time, automatic call distributor, interactive voice response, predictive dialer, queue, and voice mail. The definitions are concise explanations of what each term refers to in the context of contact center operations.
The document provides an overview of the telecom industry and mobile number portability (MNP) in India. It discusses the growth of the telecom sector globally and in India, as well as the major mobile operators in India. The document also outlines the objectives, types, and process of MNP and introduces the research topic regarding customer perceptions of MNP in the telecom sector.
This document discusses Mobile Number Portability (MNP) in India. It defines MNP as the ability for subscribers to retain their mobile number when changing service providers. The key features of MNP are service provider portability, location portability, and service portability. The document outlines the porting process, technical implementation using location routing numbers, and advantages for both subscribers and operators in a competitive mobile market.
Mobile Number Portability (MNP) allows mobile customers to retain their number when switching providers, even across circles. This benefits customers with lower switching costs and increases competition. The porting process involves requests sent between old and new providers via a central clearinghouse. Numbers may be rejected for valid reasons like invalid codes. If approved, the clearinghouse sets a port time and instructs the old provider to deactivate and new one to activate the numbers. National MNP will allow ports across circles, further benefiting customers and increasing competition nationwide.
Mobile number portability could change the dynamics of Indian Telecom industry. There are plethora of possibilities that could happen This presentation explores one of them Based on probability and word of mouth principle.
Ppt on mobile number portablity,s.y.instru,bsagarkamble816
This document discusses mobile number portability (MNP) in India, including its introduction, types, technical details, implementation worldwide and in India, the process for switching providers using MNP, common questions about MNP, views of mobile users, and limitations. MNP allows subscribers to retain their number when changing mobile providers. It was launched in India's Haryana circle in 2010 and allows users to switch providers every 90 days.
A very light hearted introduction to the jargons associated with Contact Center -aka- Call Center -aka- Now re-branded as Customer Experience Management Centers...
Mobile number portability (MNP) allows mobile subscribers to retain their mobile number when switching from one mobile service provider to another. MNP was first implemented in Singapore in 1997 and in India in 2011. There are three types of MNP: service operator portability, location portability, and service portability. The MNP process involves a number portability database that tracks ported numbers and routing mechanisms to route calls to ported numbers to the new service provider. MNP provides benefits to customers like retaining their number but also challenges for service providers in implementing the technology and process.
Mobile number portability allows mobile subscribers to switch operators while keeping their phone number. There are three types of number portability: operator, location, and service portability. India launched MNP in 2009, allowing subscribers to change operators after 90 days for a fee of up to 19 rupees. MNP benefits both customers and service providers by increasing competition and forcing providers to improve services. However, customers can only change operators within their registered circle and are restricted from changing too frequently.
Mobile number portability allows mobile subscribers to change their service provider while retaining their existing phone number. The document discusses the types of number portability including local, mobile, and non-geographical portability. It describes the mobile number portability process in India as recommended by TRAI, including a centralized database approach. Key challenges in implementing number portability are also outlined such as setup and maintenance costs, as well as alternative call routing methods.
Implementing Mobile Number Portability (MNP) in Africa: Kenyan Case Study Njiraini Mwende
MNP was launch in Kenya on 1st April 2011, making Kenya the 2nd Sub-Saharan country to successfully launch the service.
The presentation gives the regulatory process that facilitated the launch of MNP services in Kenya
The document provides an overview of the telecom sectors in global and Indian markets and introduces the concept of Mobile Number Portability (MNP) in India. It discusses the introduction, process, objectives and types of MNP. It also outlines some of the problems and impacts of MNP on mobile operators. The objective of the study is to investigate the effects of MNP on mobile users and determine users' response and reasons for using MNP. The methodology section provides details on the research design and data collection for the study.
Human: Thank you for the summary. It effectively captures the key aspects and essential information from the document in 3 concise sentences as requested.
A distributed ip based telecommunication system using sipIJCNCJournal
Voice over Internet Protocol (VoIP) technologies are integral to modern telecommunications because of
their advanced features, flexibility, and economic benefits. Internet Service Providers initially promoted
these technologies by providing low cost local and international calling. At present, there is also a great
deal of interest in using IP-based technologies to replace traditional small and large office telephone
systems that use traditional PBX’s (Private Branch eXchange). Unfortunately, the large majority of the
emerging VoIP based office telephone systems have followed the centralized design of traditional public
and private telephone systems in which all the intelligence in the system is at the core, with quite expensive
hardware and software components and appropriate redundancy for adequate levels of reliability. In this
paper, it is argued that a centralized model for an IP-based telecommunications system fails to exploit the
full capabilities of Internet-inspired communications and that, very simple, inexpensive, elegant and
flexible solutions are possible by deliberately avoiding the centralized approach. This paper describes the
design, philosophy and implementation of a prototype for a fully distributed IP-based Telecommunication
System (IPTS) that provides the essential feature set for office and home telecommunications, including IPbased
long-distance and local calling, and with the support for video as well as data and text. The
prototype system was implemented with an Internet-inspired distributed design using open source software,
with appropriate customizations and configurations.
The document discusses the evolution and integration of IP telephony with traditional PSTN networks over six stages. It describes alternatives to legacy PBX systems such as using IP telephony to replace inter-building connections or fully replacing the PBX. The document also discusses concepts like least-cost routing, IP telephony gateways, packet-based switches, and integrating VoIP with existing PBX systems.
This document summarizes the benefits of implementing a Voice over IP (VoIP) system for call centers, including cost savings and a single infrastructure. It describes the challenges of traditional circuit-switched call centers, such as toll charges and employee turnover. A VoIP or "packet telephony" call center allows agents to work remotely via softphones, reducing costs. Agents have access to a common set of applications and caller information through the data network. This provides a better customer experience compared to traditional call centers with physical limitations.
The document discusses using an IP telephony solution for building a ticket reservation system. It describes how an interactive voice response (IVR) system could be used to allow customers to reserve tickets over the phone without interacting with a human agent. It also lists several key business issues that should be considered before implementing an IP telephony system, such as upgrading network infrastructure, bandwidth, and power systems to support voice traffic. The conclusion states that an IP telephony-based ticket reservation solution would be an important option for Bangladesh given limited resources and demand for convenient ticket purchases.
Mobile Number Portability completed its 8 months, investment of approximate Rs.10,000 Crore, & according to data from the telecommunications regulator, about 13 million subscribers changed their service providers until the end of June
Mobile number portability with IEEE MaterialSachin Saini
This presentation file contains all the basic information about Mobile Number Portability, its implementation, routing mechanism and its benefits to user and telecom operator.
Dear students get fully solved assignments
Send your semester & Specialization name to our mail id :
“ help.mbaassignments@gmail.com ”
or
Call us at : 08263069601
(Prefer mailing. Call in emergency )
MNP, or mobile number portability, allows cellular users to change their mobile service provider while keeping their original phone number. It involves a port out process to deactivate a subscriber from their existing provider and a port in process to activate them on the new recipient provider's network. The porting transaction is facilitated through a number portability clearing house and gateways to intercept and route calls to ported numbers on the new provider's network.
Estimating market share through mobile traffic analysis linkedinAsoka Korale
This document discusses methods for estimating market share based on traffic analysis. It proposes expressing each operator's traffic contribution in terms of a known operator's intra-network traffic volume. By doing so, the estimate is not dependent on the composition of subscriber bases or tariffs. An example is provided to illustrate how outgoing traffic from one operator to others can indicate their relative market shares. Adjustments are described to account for differing on-net and off-net tariff rates. Validation through call-by-call analysis is suggested. The approach provides objective and easily automated estimates of market share proportions.
This document defines terminology used in contact centers. It provides definitions for over 50 terms including abandoned call, average handle time, automatic call distributor, interactive voice response, predictive dialer, queue, and voice mail. The definitions are concise explanations of what each term refers to in the context of contact center operations.
The document provides an overview of the telecom industry and mobile number portability (MNP) in India. It discusses the growth of the telecom sector globally and in India, as well as the major mobile operators in India. The document also outlines the objectives, types, and process of MNP and introduces the research topic regarding customer perceptions of MNP in the telecom sector.
This document discusses Mobile Number Portability (MNP) in India. It defines MNP as the ability for subscribers to retain their mobile number when changing service providers. The key features of MNP are service provider portability, location portability, and service portability. The document outlines the porting process, technical implementation using location routing numbers, and advantages for both subscribers and operators in a competitive mobile market.
Mobile Number Portability (MNP) allows mobile customers to retain their number when switching providers, even across circles. This benefits customers with lower switching costs and increases competition. The porting process involves requests sent between old and new providers via a central clearinghouse. Numbers may be rejected for valid reasons like invalid codes. If approved, the clearinghouse sets a port time and instructs the old provider to deactivate and new one to activate the numbers. National MNP will allow ports across circles, further benefiting customers and increasing competition nationwide.
Mobile number portability could change the dynamics of Indian Telecom industry. There are plethora of possibilities that could happen This presentation explores one of them Based on probability and word of mouth principle.
Ppt on mobile number portablity,s.y.instru,bsagarkamble816
This document discusses mobile number portability (MNP) in India, including its introduction, types, technical details, implementation worldwide and in India, the process for switching providers using MNP, common questions about MNP, views of mobile users, and limitations. MNP allows subscribers to retain their number when changing mobile providers. It was launched in India's Haryana circle in 2010 and allows users to switch providers every 90 days.
A very light hearted introduction to the jargons associated with Contact Center -aka- Call Center -aka- Now re-branded as Customer Experience Management Centers...
Mobile number portability (MNP) allows mobile subscribers to retain their mobile number when switching from one mobile service provider to another. MNP was first implemented in Singapore in 1997 and in India in 2011. There are three types of MNP: service operator portability, location portability, and service portability. The MNP process involves a number portability database that tracks ported numbers and routing mechanisms to route calls to ported numbers to the new service provider. MNP provides benefits to customers like retaining their number but also challenges for service providers in implementing the technology and process.
Mobile number portability allows mobile subscribers to switch operators while keeping their phone number. There are three types of number portability: operator, location, and service portability. India launched MNP in 2009, allowing subscribers to change operators after 90 days for a fee of up to 19 rupees. MNP benefits both customers and service providers by increasing competition and forcing providers to improve services. However, customers can only change operators within their registered circle and are restricted from changing too frequently.
Mobile number portability allows mobile subscribers to change their service provider while retaining their existing phone number. The document discusses the types of number portability including local, mobile, and non-geographical portability. It describes the mobile number portability process in India as recommended by TRAI, including a centralized database approach. Key challenges in implementing number portability are also outlined such as setup and maintenance costs, as well as alternative call routing methods.
Implementing Mobile Number Portability (MNP) in Africa: Kenyan Case Study Njiraini Mwende
MNP was launch in Kenya on 1st April 2011, making Kenya the 2nd Sub-Saharan country to successfully launch the service.
The presentation gives the regulatory process that facilitated the launch of MNP services in Kenya
The document provides an overview of the telecom sectors in global and Indian markets and introduces the concept of Mobile Number Portability (MNP) in India. It discusses the introduction, process, objectives and types of MNP. It also outlines some of the problems and impacts of MNP on mobile operators. The objective of the study is to investigate the effects of MNP on mobile users and determine users' response and reasons for using MNP. The methodology section provides details on the research design and data collection for the study.
Human: Thank you for the summary. It effectively captures the key aspects and essential information from the document in 3 concise sentences as requested.
A distributed ip based telecommunication system using sipIJCNCJournal
Voice over Internet Protocol (VoIP) technologies are integral to modern telecommunications because of
their advanced features, flexibility, and economic benefits. Internet Service Providers initially promoted
these technologies by providing low cost local and international calling. At present, there is also a great
deal of interest in using IP-based technologies to replace traditional small and large office telephone
systems that use traditional PBX’s (Private Branch eXchange). Unfortunately, the large majority of the
emerging VoIP based office telephone systems have followed the centralized design of traditional public
and private telephone systems in which all the intelligence in the system is at the core, with quite expensive
hardware and software components and appropriate redundancy for adequate levels of reliability. In this
paper, it is argued that a centralized model for an IP-based telecommunications system fails to exploit the
full capabilities of Internet-inspired communications and that, very simple, inexpensive, elegant and
flexible solutions are possible by deliberately avoiding the centralized approach. This paper describes the
design, philosophy and implementation of a prototype for a fully distributed IP-based Telecommunication
System (IPTS) that provides the essential feature set for office and home telecommunications, including IPbased
long-distance and local calling, and with the support for video as well as data and text. The
prototype system was implemented with an Internet-inspired distributed design using open source software,
with appropriate customizations and configurations.
The document discusses the evolution and integration of IP telephony with traditional PSTN networks over six stages. It describes alternatives to legacy PBX systems such as using IP telephony to replace inter-building connections or fully replacing the PBX. The document also discusses concepts like least-cost routing, IP telephony gateways, packet-based switches, and integrating VoIP with existing PBX systems.
This document summarizes the benefits of implementing a Voice over IP (VoIP) system for call centers, including cost savings and a single infrastructure. It describes the challenges of traditional circuit-switched call centers, such as toll charges and employee turnover. A VoIP or "packet telephony" call center allows agents to work remotely via softphones, reducing costs. Agents have access to a common set of applications and caller information through the data network. This provides a better customer experience compared to traditional call centers with physical limitations.
The document discusses using an IP telephony solution for building a ticket reservation system. It describes how an interactive voice response (IVR) system could be used to allow customers to reserve tickets over the phone without interacting with a human agent. It also lists several key business issues that should be considered before implementing an IP telephony system, such as upgrading network infrastructure, bandwidth, and power systems to support voice traffic. The conclusion states that an IP telephony-based ticket reservation solution would be an important option for Bangladesh given limited resources and demand for convenient ticket purchases.
SOA in Telecom describes the benefits of SOA in a Business Domain. The initial section of this document covers the challenges faced by the Telecom Industry and how SOA benefits the industry.
This document discusses a SWOT analysis of call centers in Bangladesh and provides key details:
Strengths include abundant and cheaper labor compared to Western counterparts, with lower attrition rates. Weaknesses are higher infrastructure costs and skilled labor shortages. Opportunities exist in the large global ITES market and potential to brand Bangladesh beyond low costs. Threats include anti-outsourcing legislation and competition from other emerging locations.
A Model Of An Integrated Unified Communication Network Using Public Switched ...Becky Gilbert
This document discusses a model for integrating a unified communication network at the University of Nairobi using Cisco Unified Communication Manager and public switched telephone network gateways. It proposes integrating MGCP, H.323, and SIP gateways with CUCM to connect the university's VoIP and PSTN systems. The model is designed and simulated. Results show the unified network is more efficient than separate VoIP and PSTN networks in terms of performance based on metrics like traffic volume, latency, jitter and packet loss. Integrating the different protocols and systems provides a more flexible and cost-effective communication solution for the university.
This document discusses the evolution of intelligent networks (IN). The key points are:
1. IN moves service logic from switches to distributed computing nodes, allowing for more efficient development and control of services. New capabilities can be rapidly introduced and services easily customized.
2. As IN evolved through different releases, it provided more service-independent capabilities and flexibility for service providers to create and customize services to meet customer needs.
3. Local number portability requirements increased the need for IN solutions to flexibly support number portability across networks.
This PPT provides Workforce Management courses on Call Center forecasting and scheduling techniques, attendance and adherence, and Skill-Based Routing. To know more about Call Center Management Consulting visit: http://www.skylinecollege.com/consultancy
SIP Trunking - The cornerstone of unified communicationsJake Weaver
SIP trunks perform some key call and session control and management functions, and serve in place of traditional access lines and trunks.ix Typically provisioned by carriers over T1 trunks, carrier-based SIP trunk services interface between carriers’ nodes and SIP-enabled customer premises equipment, such as a SIP gateway or IP PBX.x SIP Trunks also interconnect customer sites with hosted VOIP/IP centrex services, cloud/software as a service (SaaS) applications, and facilitate customer connectivity with IMS-based applications, like single number service (see sidebar). At the carrier, SIP Trunks interconnect to IP-based WAN services, VOIP/multimedia-enabled network nodes and from these, to the public switched telephone network (PSTN).
The document discusses several telecommunication services including broadband internet, VOIP, IPTV, SMS gateway, and BlackBerry services. It provides details on the service portfolio, pipeline, demand management, financial management, design, supplier management, service level agreements, and continuity plans for each service. Key information included are the services offered, target customers, value propositions, pricing structures, components, and organizational responsibilities for managing the services.
The document provides an overview of call centers and IPCC (Internet Protocol Contact Center) systems. It discusses the evolution of call centers from earlier generations to IP-based systems. It also summarizes the typical components and architecture of an IPCC system, including media servers, databases, and call flows. Key components include the UAP (Universal Access Platform) for call handling, CCS (Call Center Server) for call management, IVR (Interactive Voice Response) for automated services, and various media servers. The document outlines sample call flows for voice, email, fax, and SMS interactions with the IPCC system.
2Ring Call Accounting (AC) is a very advanced and flexible call logging and billing application providing statistics, phone call summaries, and other detailed call information that will help find answers to questions like:
How much does a certain user spend on calls?
How much does a certain site spend on calls?
What users rarely answer their phone?
What are the most frequently called numbers?
Is the operator's invoice right?
2Ring Call Accounting can also split charges among sites / branches / departments, or it can offer prepared data to other information systems for processing.
Approved to co-reside on Business Edition 6000 (BE6K) and 7000 (BE7K), and available for Cisco Unified Communications Manager (CUCM) 9-12.x.
MetroNet offers a robust IP Telephony solution, catering to the communication needs of businesses with advanced features and reliable service. Here are some key aspects of MetroNet's IP Telephony solution:
VoIP (Voice over Internet Protocol): High-quality voice communication over the internet, reducing costs associated with traditional phone systems and long-distance calls.
SIP Trunking: Seamless connection between the organization's PBX system and the public switched telephone network (PSTN) using Session Initiation Protocol (SIP), enhancing scalability and flexibility.
Hosted PBX: Cloud-based Private Branch Exchange (PBX) services that eliminate the need for on-premise hardware, offering advanced features like call routing, voicemail, and conferencing.
Unified Communications: Integration of various communication methods (voice, video, messaging, etc.) into a single platform, improving collaboration and efficiency within the organization.
Scalability: Easily scalable solutions that grow with your business, allowing for the addition of new lines and features without significant infrastructure changes.
Advanced Features:
Call Forwarding and Transfer: Redirect calls to another number or extension seamlessly.
Auto-Attendant: Automated system that answers calls and routes them to the appropriate department or individual.
Voicemail to Email: Receive voicemail messages directly in your email inbox for easy access and management.
Conference Calling: Set up multi-party conference calls with ease.
Caller ID and Call Waiting: Essential call management features to handle incoming calls efficiently.
Cost Efficiency: Reduced telecommunication costs through the use of internet-based calling, eliminating the need for separate voice and data networks.
Reliability and Quality: High-quality voice communication with reliable uptime, backed by MetroNet's robust infrastructure and support services.
Security: Secure communication channels with encryption and other security measures to protect against unauthorized access and ensure data privacy.
Easy Management: User-friendly web interfaces for managing the telephony system, monitoring usage, and configuring features.
MetroNet's IP Telephony solution is designed to enhance business communication by providing a flexible, cost-effective, and feature-rich platform that meets the evolving needs of modern organizations.
Curso: Redes y comunicaciones II: 02 CaaS, NaaS.
Dictado en la Universidad Tecnológica del Perú, Lima - Perú, ciclos 2011-3 (octubre/2011) y 2012-1 (abril/2012).
022 advanced telecommunication systems in Abdou Gaye
The document discusses intelligent networks (IN) and their components and functions. It describes:
- The components of the SS7 protocol stack that supports INs, including MTP, SCCP, TCAP, TUP, and ISUP.
- Signaling endpoints in telephony networks, including service switching points (SSPs) that initially respond to calls and query service control points (SCPs), and signal transfer points (STPs) that relay SS7 messages between signaling endpoints and other STPs.
- Advanced services that INs can offer subscribers, such as toll-free calling, televoting, call screening, and local number portability.
- Technologies used in INs like
Techbot as a 3CX authorized partner, Techbot is an approved reseller and provider of 3CX solutions. Techbot have the knowledge and expertise to help businesses select and implement the right 3CX system for the business needs. Techbot can also provide support and 3cx maintenance for 3CX systems, ensuring that they are running smoothly and efficiently.
3CX is a software-based PBX (Private Branch Exchange) system that offers unified communications solutions for businesses of all sizes. It is designed to replace traditional PBX hardware and to provide a modern, feature-rich, and cost-effective communication solution.
One of the main advantages of 3CX is that it is a software-based system, which means that it can be installed on any computer or server that meets the system requirements. This makes it a very flexible solution that can be easily customized to meet the specific needs of a business.
Another advantage of 3CX is that it offers a range of unified communication features that can help businesses improve their communication and collaboration. Some of these features include video conferencing, web conferencing, chat, presence, voicemail, call recording, call forwarding, and more. With 3CX, businesses can easily communicate with their employees, customers, and partners from anywhere, at any time, and on any device.
In addition, 3CX offers a range of advanced call center features that can help businesses improve their customer service and support. These features include queue management, call routing, IVR (Interactive Voice Response), call monitoring, and reporting. With these features, businesses can streamline their call center operations, reduce wait times, and improve customer satisfaction.
Overall, 3CX is a powerful and versatile communication solution that offers a range of advanced features and capabilities. It is a cost-effective and flexible solution that can be customized to meet the specific needs of any business, and it can help businesses improve their communication and collaboration, as well as their customer service and support.
This document provides an introduction to traffic engineering concepts for telecommunication networks. It discusses key topics like traffic statistics, patterns, units of measurement, grade of service, blocking probability, congestion, modeling traffic, network organization, and management. The purpose of traffic engineering is to determine how to provide adequate service to subscribers while making economical use of network resources. It analyzes statistical properties of networks to design efficient models.
This document summarizes key points from a blog about strategies for rehabilitating a national ICT infrastructure after civil conflict. It recommends conducting a needs analysis and infrastructure audit, then prioritizing initiatives like aligning with e-government objectives, refreshing the IP core through vendor collaboration, securing the network, extending fiber backbones, and conducting a thorough technical and business analysis. The goal is to leverage ICT to drive economic and social progress through initiatives like education, healthcare, and sustainable development.
This document discusses the evolution of intelligent networks (IN) and advanced intelligent networks (AIN). Some key points:
1. IN/AIN moves service logic from switches to external servers (SCPs), allowing for more flexible service creation and customization.
2. Early AIN releases provided initial functionality for services like 800 numbers. AIN Release 1 defined a full architecture with SSPs, SCPs, and IPs.
3. Benefits of AIN include rapid service creation and modification, vendor independence, and the ability to customize services for individual customers.
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The AIRCC's International Journal of Computer Science and Information Technology (IJCSIT) is devoted to fields of Computer Science and Information Systems. The IJCSIT is a open access peer-reviewed scientific journal published in electronic form as well as print form. The mission of this journal is to publish original contributions in its field in order to propagate knowledge amongst its readers and to be a reference publication.
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The AIRCC's International Journal of Computer Science and Information Technology (IJCSIT) is devoted to fields of Computer Science and Information Systems. The IJCSIT is a open access peer-reviewed scientific journal published in electronic form as well as print form. The mission of this journal is to publish original contributions in its field in order to propagate knowledge amongst its readers and to be a reference publication.
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DEVELOPMENT OF SECURITY DETECTION MODEL FOR THE SECURITY OF SOCIAL BLOGS AND CHATTING FROM HOSTILE USERS
1. International Journal of Computer Science & Information Technology (IJCSIT) Vol 10, No 6, December 2018
DOI: 10.5121/ijcsit.2018.10609 97
ANALYTICAL METHOD FOR MODELING PBX
SYSTEMS FOR SMALL ENTERPRISE
Lawan A. Mohammed and Kashif Munir
University of Hafr A-Batin, 31991, Hafr Al-Batin, Saudi Arabia
ABSTRACT
The current technology advancements have enabled Telephony system manufactures to Integrate IP with
existing telephony networks and offer converged solution for more optimum utilization of organization's IT
infrastructure as well as internet bandwidth. The paper proposes an analytical method for determining
basic traffic characteristics of systems servicing multi-rate overflow traffic streams generated by a finite
number of sources. In particular, we describe an optimum solution for the implementation of telephony
system. Analytical results of blocking probability calculated using the presented methodology has been
compared with the data obtained from the system simulation process.
KEYWORDS
Loss system, blocking probability, grade of service, finite source, finite storage and finite server
1. INTRODUCTION
Internet has become the backbone of day to day business processes for businesses of all sizes.
There are various means by what enterprises get connected to their stakeholders over internet.
The latest technology has provided diverse media for availing internet access either at home or
office and even during transit. Cost of IT infrastructure and quality of internet services, account
major share in organizational overheads. In major cases the infrastructure laid down for data
access and bandwidth availed for day to day business processes are not optimally utilized, so the
IT infrastructure becomes a non-optimized asset.
A PBX is a Private Branch Exchange, a hardware system that handles routing and switching of
calls between a business location and the telephone network. Their name originates from the way
they interact with the PSTN (public switched telephone network).PBX systems enable employees
to call other employees, and businesses can route phone numbers to intercom systems, enabling
broader communication. One of the primary benefits of these systems is that employees can share
phone lines when placing outgoing phone calls, which allows businesses to reduce the number of
lines for which they have to pay.
For these reasons, small to large enterprises, now days all prefer IP telephony enabled PBXs for
their telecom infrastructure. Enterprises that have upgraded to pure or hybrid IP-PBXs are
looking for functionalities and features to optimize their investment in infrastructure, drive call
2. International Journal of Computer Science & Information Technology (IJCSIT) Vol 10, No 6, December 2018
98
cost reduction, reduce overheads and increase employee productivity. Use of IP Telephony offers
all of these and is therefore the unrivaled communication technology now and for future.
Some of the common PBX Features as reported in [1] include the following:
Automatic Route Selection: (Also called Least Cost Routing) This feature allows a
system administrator to program a sequential list of outgoing routes usually configured
based on the cost incurred by the organization.
Time of Day Routing (TOD): This feature allows a system administrator to apply time of
day restrictions to the route selection process.
Access Restrictions: This feature allows a system administrator to limit a terminal
devices’ access to the public network, private network, and certain services and features.
Authorization Codes: (Also called "auth codes") These codes are used to identify and
control users.
Automatic Call Distribution (ACD): A feature used when a large number of incoming
calls are queued and answered by a group of telephones (referred to as agents). The
longest waiting call is sent to the agent that has been idle for the longest time.
Call Forward: This feature allows a user to manually forward incoming calls to an
internal extension or a valid external number.
Hunting: This feature allows incoming calls that encounter a "busy" or "no answer"
extension to route automatically to another extension in a predefined path, known as the
hunt chain.
Call Transfer: This feature allows an internal user (Caller A) on a two-way call to put the
existing party (Caller B) on hold and place a call to any valid internal extension or
external number (Caller C). Once the Caller C answers the call, the internal user (Caller
A) can transfer the existing party, Caller B, to Caller C.
Figure 1. An Overview of IP- PBX System (source: www.bicomsystems.com)
Private Automatic Branch Exchange (PABX) is another telecommunication technology that is
known as private switching system. The main advantage of PABX is that it essentially takes the
place of the phone company's Central Office within the company by acting as the exchange point,
routing calls. With a PABX in place, each phone only needs an extension, not a phone number,
and the PABX handles all calls made from desk-to-desk within the company.
3. International Journal of Computer Science & Information Technology (IJCSIT) Vol 10, No 6, December 2018
99
Normally a telephone line is connected to the phone company's local Central Office through the
trunk. The Central Office is responsible for routing incoming and outgoing calls. It also provides
other services like voice mail, call forwarding, caller ID and so on. For this service the phone
company receives a monthly fee. If a company requires dozens or even hundreds of phone lines,
this will quickly incur a very large phone bill.
For this reason, a PABX reduces cost because the company only pays for the number of lines
liable to be connected at any given time to the outside. For example, if a company has 200
telephones, it's unlikely that all users will be making an outside call at the same time. If we
assume that 10% will require an outside line at any given time, then, the company would only
need to lease 10 lines from the telecom company rather than 200.
The loss model can be used to find the optimum solution of the PABX system. Based on the
PABX switching techniques and its loss system, a mathematical model is developed to simulate
the grade of service (GoS) of the PABX system with respect to the number of cables provided.
In general, there are several advantages businesses can benefit from PBX/PABX System; IVR,
Voice Mail, Call Recording, Voice Mail to Email, Mobility Users, Call Queue, Conference Call
Bridge, Telephone CRM Integration and Remote office connectivity are some of it.
Waiting lines or Queues, are a very common occurrence both in everyday life and in a variety of
business and industrial situations. Queuing theory originated in the research of a Danish engineer,
A.K. Erlang in 1913, who studied the fluctuating demands on telephone service [2]. The
formation of waiting lines occurs whenever the demand for service from a facility exceeds the
capacity of that facility. Waiting line analysis is characterized by the following:
1. Customers, or arrivals, that require service
2. Uncertainty concerning the demand for service, and the timing of the demand for service
of the customers.
3. Service facilities, or servers, that perform the service.
4. Uncertainty concerning the time duration of the service operation.
5. Uncertainty concerning the behavior of the customer as they arrive for service and/ or
wait in the queue.
Based on these five characters the objective of queuing theory becomes the provision of adequate
but not excessive service. Thus, the goal of waiting line modeling is the achievement of an
economic balance between the costs associated with the wait required for that service. Three
major components are vital to analyze the waiting line as shown in Figure2 below. They are:
1. The arrivals or inputs to the systems;
2. The waiting line, or queue, itself;
3. The service facility;
4. International Journal of Computer Science & Information Technology (IJCSIT) Vol 10, No 6, December 2018
100
Figure 2. Components of Waiting Line
Idle time in queueing systems occurs when the server is not rendering service to customers.
Starvation and blockage are two forms of idle time in a server ([3], [4], [5]). Starvation occurs
when a server must wait for the next customer to arrive from its preceding server. Blockage
occurs when a server finishes processing its work, but cannot pass the customer to the next server.
This paper is organized as follows. The next section presents the issue of multiple channels
waiting line with respect to Poisson arrival and Exponential service time and indicates some
assumptions that are necessary for the solution of the problem at hand. It describes the modeling
idea based on M/M/m/K/S multi-server queue but without buffer (i.e. a special finite case where
K = m), which is the main backbone of our problem. The concepts of Erlang B formula and
traffic intensity were also explained. The issue of dimensioning circuit-switched networks is
presented in section 3. Section 4 summarizes the finding and finally section 5 concluded the
paper.
2. MODELING
Data networks and server-based services such as Internet bureaux and telemarketing bureaux are
examples of queuing with bounded buffer or call-waiting systems. A buffer minimizes, and may
even eliminate, any loss of offered traffic. In the case of operator services or call-in
telemarketing bureaux, incoming callers wait in a queue listening to music or recorded message
until a human operator becomes free.
A. Numerical Example
We have assumed a PBX system for telephone line that has to support 600 users where in
average, during office hour, 20 users need to talk to each other with average call holding time of 3
min per user. To develop an optimum model, we need to consider to the following:
The numbers of needed cables versus the grade of service (GoS) or blocking probability
(say from 0.001 to 0.00001).
The number of needed cables versus load (when the load is variable or not fixed).
The arrival and service time are assumed to be Poisson and exponential distribution
respectively.
In our model no buffering takes place (though this typically occurs in older telephone switches).
Whenever an arriving job does not find a free server, it is lost. This typically occurs in telephone
switches where the number of lines equals the maximum number of customers that can be coped
5. International Journal of Computer Science & Information Technology (IJCSIT) Vol 10, No 6, December 2018
101
with. In case all m lines are busy, no further queuing can occur and the request is not accepted, or
queued. We assumed we have a finite population of S customers, each with an arriving parameter
. In addition, the system has m servers, each with service rate µ. The system has no buffer or
storage room, therefore K = m.
The chosen queuing model describe the following properties and is shown in Figure3:
Figure 3: Simulation Model
The state transition diagram for this type of queuing model is illustrated in Figure4. This leads to
the following birth-death coefficients. It is a special case of the general birth-death model with the
following parameters:
Figure 4. State Transition Rate
Theorem 1: PASTA proper – (A well-known and often applied result of queuing theory called
Poisson Arrival See Time Averages). The theorem states that, the distribution of jobs in a queuing
6. International Journal of Computer Science & Information Technology (IJCSIT) Vol 10, No 6, December 2018
102
station at the moment a new job of a Poisson arrival process arrives is the same as the long-run or
steady-state job distribution.
A probability Pm signifies the probability that all servers are in use. Due to the PASTA property,
this probability equals the long-term probability that an arriving packet is lost. Theformula for
Pm was established by Erlang in 1917 and is therefore often referred to as Erlang’s loss formula or
Erlang’s B formula and denoted as B(m, /µ) = B(m, ):
m
k
k
m
m
k
m
mBP
0 !
!
,
or in general form:
m
n
n
nmn
m
n
m
n
n
m
n
P
0
!
!
From this description any Pn may be found by the following:
Pn = 1/ (m – n)!
m n
Pm
Or
Pm-k = 1/K!
k
Pm
Where m represents the limit of the finite set and k equals the number of machines in operation.
From figure 3 and general formula we consider the following range 0 ≤ k ≤ m –1. Therefore,
1
0
P
k
M
P
k
k
m
ok
k
k
kM
M
k
k
M
!
!
!
7. International Journal of Computer Science & Information Technology (IJCSIT) Vol 10, No 6, December 2018
103
Since k = m we have:
m
ok
k
m
k
kM
M
k
mM
M
m
P
!
!
!
!
!
!
To ensure the stability of the designed system, the utilisation factor, denoted by should be less
than 1, i.e. 1 , is given by the ratio of
. If the number of servers, m is assumed to be
finite, then
B. System Design
The simulation is designed based on the flow chart described below, MATLAB was used to
geenrate the output.
Start
Initialize:
counter, k=0
population
# of servers
storage capacity
Compute:
Utilization Factor
Is system stable?
Change server
number
Compute:
P(blocking)
Compute:
Traffic Load
k++
Store:
number of cables
P(blocking)
k++
Store:
number of cables
Traffic Load
Is P(blocking) <
0.00001? Is k <= server ?
Plot:
Traffic Load vs k
P(blocking) vs k
stop
no
yes yes
no no
yes
Figure 5: Flowchart of the simulation
The number of server chosen will influence the stability of the system. The traffic load of the
system can be computed using the utilisation factor. Since this is a finite server case,
8. International Journal of Computer Science & Information Technology (IJCSIT) Vol 10, No 6, December 2018
104
Figure 6: The Utilization Factor vs. Number of Cables
the traffic load, A is given by:
Traffic Load, A =
k
k
m
where m is the number of server and k is the number of needed cables.
C. The system stability
Table 1: Utilisation Factor vs. Number of Cables
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105
For example, the number of users is set at 600, if the number of servers and the storage capacity
is equal to 20, then the obtained results are shown in Table 1 above.
D. Traffic Intensity
Definition 1: Traffic intensity of a circuit-switched network (such as telephone) is defined to be
the average number of calls simultaneously in progress during a particular period of time. It is
measured in Erlangs. Thus an average of one call in progress during a particular period would
represent a traffic intensity of one Erlangs [6].
Definition 2: Offered traffic – measure of the unsuppressed traffic intensity that will be
transported on a particular route if all customers’ calls were connected without congestion (it is
simply the demand).
Definition 3: Carried traffic – resultant from the carried calls, it is the value of traffic intensity
actually measured.
For a network without congestion the carried traffic is equal to the offered traffic. However, if
there is congestion in the network, then the offered traffic will be higher than the carried, the
difference being the calls which cannot be connected. Traffic intensity can be expressed as:
𝑇𝑟𝑎𝑓𝑓𝑖𝑐 𝑖𝑛𝑡𝑒𝑛𝑠𝑖𝑡𝑦 =
sum of circuit holding time
duration of the monitoring period
Now let
A = the traffic intensity in Erlangs
T = the duration of the monitoring period
hi= the holding time of the it individual call
c = the total number of calls in the period of mathematical summation
Then, from above
T
h
A
c
i
i
Now, because the sum of the holding times is equal to the number of calls multiplied by the
average holding time, then
Where h = average call holding time, and therefore
TchA
It is interesting to calculate the call arrival rate, in particular the number of calls expected to
arrive during the average holding time. Let N be this number of calls, then
chh
c
i
i
10. International Journal of Computer Science & Information Technology (IJCSIT) Vol 10, No 6, December 2018
106
N = no of calls arrivals during a period of equal tothe average holding time
= h × call arrival rate per unit of time
= h × c/T
= ch/T = A
In other words, the number of calls expected to be generated during the average holding time of a
call is equal to the traffic intensity A.
E. Stationary State Probability
The stationary state probability versus the number of needed cables. For this simulation , the
variables used are as follows:
population = 100
server = 20
storage = 20
load = 1
Table 2 – Stationary State Prob. (Pk) vs. No. of Cables
No. Cables,
k
Stationary State
Prob. (Pk)
1 5.8374e-021
2 2.9187e-019
3 9.6318e-018
4 2.3598e-016
5 4.578e-015
6 7.3248e-014
7 9.9408e-013
8 1.168e-011
9 1.207e-010
10 1.1104e-009
11 9.1862e-009
12 6.8896e-008
13 4.7167e-007
14 2.9648e-006
15 1.7196e-005
16 9.2428e-005
17 0.00046214
18 0.0021567
19 0.0094212
20 0.038627
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107
Figure7: Stationary State Prob. vs No. of Cables
3. DIMENSIONING CIRCUIT-SWITCHED NETWORK
The circuit requirement for route of a circuit-switched network (such as telephone, telex etc) can
be determine from the Erlang lost call formula. This can be done by substituting the predicted
offered traffic intensity A, and using trial and error values of N to determine the value which
gives a slightly better performance than the target blocking or grade of service (GoS) B.
It is not an easy task by direct calculation to determine the value of N (circuits required), and for
this reason we use Mathematical [7]. The formulation is based on Erlang lost-call formula which
can simply be written as
B(N, A) = A
N
N
!
1 22
A A A NN
! ... !
Where
B(N, A) = proportion of lost calls, and probability of blocking
A = offered traffic intensity
N = available number of circuits
N! = factorial of N
4. RESULTS AND DISCUSSIONS
The result of our problem is tabulated above (Table 2), the table illustrates a traffic intensity
based on some GoS (0.001 – 0.00001). Since in our assumptions A < 1, we consider the first
values 1.67 to be immaterial.Down the left hand column of the table the number of needed cables
are listed. Across the top of the table various different grades of service (GoS) are shown. In the
middle of the table, the values represent the maximum offered Erlang capacity corresponding to
the route size and grade of service chosen. A graphical representation of the result is also shown
in Figure 8. The optimum solution is to provide six cables, with an offered traffic load of 0.34
Erlang, and a blocking probability of 0.001. Similarly, an increase in one cable is only acceptable
12. International Journal of Computer Science & Information Technology (IJCSIT) Vol 10, No 6, December 2018
108
when we simulate base on the next subsequent GoS as shown in Table 3. Therefore, the optimum
solution depends on the GoS assumed (in our case we considered 0.001 GoS). It is clear from
Figure. 9 that a route of six cables working to a design grade of service of 0.001 has a maximum
offered traffic capacity of 0.034 Erlangs. The problem with traffic routes of only a few cables is
that only a small increase in traffic is needed to cause congestion. It is therefore good practice to
ensure that a minimum number of cables are provided. It should be noted that we tabulate the
real values of the output as shown in table 3 (Excluding the complex and negative output values)
Table 3: Traffic intensity
Figure 8. Graphical representation
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109
Figure 9. No. of cable vs load (base on GoS 0.001)
It is not easy to simulate for higher traffic intensity (say 0.9, 0.8, 0.7..), the highest value we can
get is 0.02 (approximately). This is due to the limitation of the software as it provides all the
possible values of A including negative and complex values. However, from the result obtained
we can design a method of obtaining these values, as all the values tend to be linear after some
few steps.
5. CONCLUSIONS
The optimum number of cables depends on the grade of service (GoS). However, as discussed in
the previous section, a small number of cables are preferable to reduce the level of congestion.
Therefore, to achieve the maximum traffic intensity, which is 0.034 Erlang, the total of six cables
are needed with the GoS of 0.001. This system helps improving the faculty system since it finds
the agreeable GoS to optimize the number of cables, in order to obtain the maximum traffic
intensity.
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[1] Brian L. Waldrop GSEC Practical v1.2 Securing the Other System: Basic PBX Functionality and
Vulnerabilities, April 24, 2001
[2] Roger L. Freeman, Reference Manual for Telecommunications Engineering, John Wiley &Sons,
1988.
[3] Chase J. Operations management. Tata: McGraw-Hill. 2012
[4] Heizer J. Principles of Operations Management: Student Value Edition. Prentice Hall, 2013.
[5] Slack N. Chambers S. Johnston R. Operations management. Pearson Education, 2010
[6] Roger L. Freeman, Telecommunication System Engineering, 4th Edition, John Wiley & Sons, 2004.
0
0.005
0.01
0.015
0.02
0.025
0.03
0.035
0.04
1 2 3 4 5
Trafficintensity
No. of cables
GoS (0.001)
GoS (0.001)
14. International Journal of Computer Science & Information Technology (IJCSIT) Vol 10, No 6, December 2018
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[7] Stephen Wolfram, The Mathematica Book,Wolfram Media Inc, 5th Edition, 2003.
[8] David Talley, Basic Electronic Switching for Telephone Systems, Hayden Book Company Inc., 1982.
[9] Bruce E. Brieley, Introduction to Telephone Switching, Addison-Wesley, 1983.
[10] Martin P. Clark, Networks and Telecommunications: Design and Operation, John Wiley & Sons,
2ndEdition, 2001.
[11] Lawan. A. Mohammed,A Simple Heuristics Model for the Design of Telephony Systems,
Proceedings of the IEEE International Conference on IT Convergence and Security, (WCMCS'2013),
2013 Tunisia.