Relevant bibliographies by topics / PBX Asterisk

Contents

  1. Journal articles
  2. Dissertations / Theses
  3. Books
  4. Book chapters
  5. Conference papers

Academic literature on the topic 'PBX Asterisk'

Author: Grafiati
Published: 4 June 2021
Last updated: 4 February 2022

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Journal articles on the topic "PBX Asterisk"

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Maar, Michal, Julia Sitarova, and Milos Orgon. "Enterprise network with software Asterisk PBX based on the PLC technology." International Journal of Advances in Telecommunications, Electrotechnics, Signals and Systems 6, no. 1 (January 20, 2017): 1. http://dx.doi.org/10.11601/ijates.v6i1.187.

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This article presents the software Asterisk PBX solution design in enterprise PLC network (Power Line Communication). The description of the installation and configuration of software Asterisk PBX is involved in the design. The secure interconnection of two enterprise PLC network is implemented via the telecommunication tunnel with security grant using the Cisco routers. The connection between two Asterisk PBXs is designed in context of the establishment of the tunnel. The subject of the article is also cross/connection of exchanges Asterisk PBX and hardware PBX - IP Panasonic PBX K-NS500.
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Nuño, Pelayo, Carla Suárez, Eva Suárez, Francisco G. Bulnes, Francisco J. delaCalle, and Juan Carlos Granda. "A Diagnosis and Hardening Platform for an Asterisk VoIP PBX." Security and Communication Networks 2020 (December 29, 2020): 1–14. http://dx.doi.org/10.1155/2020/8853625.

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Voice over IP (VoIP) is a set of software and hardware technologies used for making voice calls over the Internet. VoIP has been massively deployed in corporative environments since voice and data network convergence enables unified communication services while reducing costs. The main component of a VoIP network infrastructure is the private branch exchange (PBX). Nowadays, Asterisk is the most widespread PBX deployed within corporations due to its open access technology, along with its modular and flexible design. The configuration of PBX systems usually relies on multiple configuration files composed of a vast number of parameters that may have an impact on the security of the system. Therefore, the setup of such systems tends to be complicated and prone to errors and usually requires highly specialized human intervention. In this research, a diagnosis platform for discovering vulnerabilities and security breaches in the configuration of an Asterisk PBX is presented. The proposed platform performs both reactive and proactive actions in order to reconfigure and harden an Asterisk PBX. Firstly, the platform reacts after certain events by modifying the configuration of the Asterisk PBX in order to mitigate risks. Secondly, the platform performs several on-demand assessments that also reconfigure the Asterisk PBX to improve overall security. Finally, the functionality of the platform is easily extensible and highly customizable. Extensive tests have been carried out to assess the security and performance of the Asterisk PBX when facing attacks. Results show that the security of the platform increases, avoiding performance degradation when using the proposed platform.
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Masudur Rahman, Mohammad, and Nafish Sarwar Islam. "VoIP Implementation Using Asterisk PBX." IOSR Journal of Business and Management 15, no. 6 (2014): 47–53. http://dx.doi.org/10.9790/487x-1564753.

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Ravonimanantsoa, N. Manda Vy, and P. Auguste Randriamitantsoa. "Comparison of the Consumption of Resources between HTTP and SIP." Advanced Engineering Forum 1 (September 2011): 330–32. http://dx.doi.org/10.4028/www.scientific.net/aef.1.330.

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Currently, the development of research around VoIP experiences a tremendous growth. In the community of open source Asterisk represents a reliable alternative for a lower cost solution. In this same community as the SIP protocol is a supplement to the more asterisk PBX. To share the benefits claimed by proponents of free software co-existence with other Asterisk server is not yet proven. In this context this paper we show a comparison of the use of simplified resource material for the apache server using the HTTP protocol and server that uses the asterisk SIP.
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Gohel, Chirag K., and Kamaljit I. Lakhtaria. "Implement VoIP Based IP Telephony with Open Source Asterisk Architecture." International Journal of Interdisciplinary Telecommunications and Networking 2, no. 1 (January 2010): 1–11. http://dx.doi.org/10.4018/jitn.2010010101.

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Asterisk is a leading open source telephony software/system, easily implemented over intranet and internet. Asterisk empowers developers and integrators to create advanced communication solutions. An Asterisk system is a low cost type of a traditional PBX system. Any phone controlled by an Asterisk system can call a VoIP or analog phone controlled or managed by a traditional telephone system or by Asterisk telephone system. In this paper, the authors focus on the deployment and testing of various Open Source Asterisk Services in an enterprise level communication system. Selected services are listed in this paper that can be used to implement a telephone system with good Quality of Services (QoS) and good Quality of Experience (QoE) from the personal user to enterprise level users.
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AlShemmary, Ebtesam Najim, and Bahaa Qasim Al-Musawi. "Low Cost VoIP Architecture Using Open Source Software Component in Tertiary Institutions." INTERNATIONAL JOURNAL OF COMPUTERS & TECHNOLOGY 3, no. 1 (August 1, 2012): 11–14. http://dx.doi.org/10.24297/ijct.v3i1a.2721.

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Governments and their agencies are often challenged by high cost and flexible telephonic, Web based data services. Emerging technologies, such as those of Voice over Internet Protocol (VoIP) that allow convergent systems where voice and Web technologies can utilize the same network to provide both services, can be used to improve such services. This paper describe VoIP system for the enterprise network (e.g. company, university) that have been developed based on Asterisk which is a kind of open source software to implement IP-PBX system. Through the development and evaluation, we have confirmed that VoIP system based on Asterisk is very powerful as a whole and most PBX functions to be required for the enterprise network can be realized. Interesting findings include that the University of Kufa has a potential to implement the project. By connecting multiple Asterisk servers located in different sites based on IAX2, large scale enterprise network can be developed. Since the software recommended for installation is open source, the project could be used as a source of valuable information by students who specialize in real-time multi-media systems.
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Meidi, Akri. "PERANCANGAN CALL CENTER INBOUND MENGGUNAKAN IP PBX ASTERISK DI UNIVERSITAS MERCU BUANA." Jurnal Ilmu Teknik dan Komputer 3, no. 1 (January 14, 2019): 63. http://dx.doi.org/10.22441/jitkom.2020.v3.i1.009.

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Penyebaran informasi merupakan suatu hal yang sangat penting seiring dengan kemajuan teknologi dan informasi. Tidak mengherankan jika setiap individu menginginkan agar informasi tersebut dapat diakses dengan cepat, tepat dan akurat. Hal ini juga terjadi di lingkungan universitas Mercu Buana dimana tuntutan ketersediaan informasi sepanjang waktu (24 jam sehari, 7 hari seminggu) menjadi kendala tersendiri bagi kampus. Call center bisa menjadi salah satu solusi untuk mengatasi masalah tersebut. Dengan kemudahan single nomor akses, call center bisa digunakan kapanpun dan dimanapun untuk mendapatkan informasi kampus. Call center inbound dirancang dengan tujuan menerima panggilan masuk dari user kemudian diteruskan ke bagian terkait sesuai dengan keinginan penelpon. Pada penelitian ini dirancang suatu sistem call center inbound dengan IVR (Interaktif Voice Response) untuk merespon penelpon secara cepat, kemudian memberikan informasi yang tersedia di sistem IVR atau mendistribusikan penelpon ke skill agent/penerima telepon yang sesuai. Semua aktifitas call dapat diukur dan dimonitoring secara realtime ataupun history call sehingga memudahkan dalam perekapan laporan.
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Bezuhlaia, N. V., M. A. Bezuhlyi, A. E. Doroshenko, and A. V. Levychev. "Methodological support of the class administration course telephone exchanges IP PBX Asterisk." Electronics and Communications 17, no. 1 (May 28, 2012): 53–57. http://dx.doi.org/10.20535/2312-1807.2012.17.1.221697.

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Odjidja, Euclid, Salah Kabanda, William Agangiba, and Richard Annan. "Wireless Enabled Voice over Internet Protocol (VoIP) Network Application Using Asterisk PBX." EAI Endorsed Transactions on Internet of Things 4, no. 15 (December 21, 2018): 156717. http://dx.doi.org/10.4108/eai.5-3-2019.156717.

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López and Pérez. "Integration of Asterisk IP-PBX with ESP32 Embedded System for Remote Code Execution." Proceedings 21, no. 1 (August 5, 2019): 38. http://dx.doi.org/10.3390/proceedings2019021038.

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This paper explains the design and construction of a platform that implements the ESP32 embedded system and connects it to a telephone asterisk plant, to exchange data on both sides, commands sent from a telephone to the esp32 and make calls from an order of sending from a digital input of esp32. It is a low-cost device that can be implemented through the use of Wi-Fi, and as a use in the industry, it has a role in analogue communication in buildings, for example.
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Dissertations / Theses on the topic "PBX Asterisk"

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Vlk, Bronislav. "Možnosti videokonferencí v PBX Asterisk." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2012. http://www.nusl.cz/ntk/nusl-219738.

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This thesis deals with the possibilities of video conferencing in Asterisk PBX and their use in practice. They also described the contingencies and how its configuration. Particular attention is paid to the protocols SIP, IAX and H.323, which are described in one of the chapters. The thesis was created by the Asterisk PBX, which demonstrates cooperation with videoconferencing clients. The thesis describes the configuration files so that the central set. Conclusion the work assesses the use of codecs for different clients.
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Benýšek, Jiří. "Vazba GSM modemu na PBX Asterisk." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2010. http://www.nusl.cz/ntk/nusl-218255.

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Short Message Service (shortly SMS) is the most widely used type of communication systems. The main advantages are that allow a fast exchange of messages between devices, a very good availability through GSM and a reasonable price. Nowadays the SMS service support has expanded to include other technologies such as a service of the information navigation and the remote connection. The master‘s thesis concentrates on the Short Message Service, deals with basic principles and statements using by this service. The topic of the thesis is software PBX Asterisk and its possibility of SMS implementation, especially verification of SMS processing goes through the PSTN. After the basic introduction the master‘s work deals with the installation and configuration of the server. The main focus is on an installation of the operating system with an additional pack including necessary libraries and modules for a correct working of the server. The following section is paying attention to the Asterisk server configuration, especially a hardware card installation which is necessary for a connection with analog telephones, done by Bluetooth connections, set up user’s profiles of the SIP protocol and create a dial plan. This is followed by a verification of SMS option of the implementation and communication with GSM modem which is used as a gate for an exchange SMS between PSTN and GSM network. The last chapter of this master‘s thesis comes with the aimed results.
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Bednář, Vít. "Implementace protokolu SIP v PBX Asterisk." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2017. http://www.nusl.cz/ntk/nusl-317016.

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The thesis compares native SIP stack with PJSIP stack in the open source telephone private branch exchange (PBX) Asterisk. First, there are described both SIP protocol and Asterisk application. Subsequently, the architecture, new function support and the stacks setting possibilities are explored. For different exchange scenarios several commented configuration files are presented. The stacks are tested using Spirent TestCenter C1 software thereafter. In conclusion, selected properties are assessed and new PJSIP stack benefits are summarized. In addition, the laboratory assignment is attached.
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Guerra, Carlos Humberto Martins Chagas. "Gateway SIP - Asterisk." Master's thesis, Universidade de Évora, 2012. http://hdl.handle.net/10174/15430.

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Esta tese consiste num estudo realizado sobre a tecnologia VoIP (Voice over IP) com aplicação dentro da área dos PBX(Private Branch eXchange) e Gateways VoIP Open Source. Em primeiro lugar foram abordados os conceitos, requisitos e elementos associados a esta tecnologia bem como a sua interligação com outras tecnologias de comunicação como a PSTN (Public Switched Telephone Network) , ISDN (Integrated Services Digital Network) ou GSM (Groupe Special Mobile).Al ém disso, foi efectuado um estudo teórico e prático sobre o software Open Source Asterisk, tendo com objectivo explorar o seu modo de funcionamento e funcionalidades disponíveis, passiveis de serem utilizadas em ambiente empresarial. Por fim, foi desenvolvida uma solução assente neste tipo de tecnologia na empresa Clidis - Laboratório de Análises Clinicas de Sines, onde ficou provado que a implementação de PBX/Gateways VoIP através de software Open Source e uma alternativa viável às reais necessidades de comunicação das empresas; ABSTRACT: This thesis is a study on VoIP (Voice over IP) application within the area of PBX (Private Branch eXchange) and VoIP Open Source Gateways. Firstly were addressed the concepts, requirements and elements associated with this technology and its interconnection with other communications technologies such as PSTN (Public Switched Telephone Network), ISDN (Integrated Services Digital Network) or GSM (Groupe Special Mobile). Furthermore, a study was carried out on the theoretical and practical Asterisk Open Source software, with the aim to explore its operation and features available, liable to be used in business environment. Finally, we developed in the company Clidis (Laboratory of Clinical Analyses in Sines) a solution based on this technology, where it was proved that the implementation of PBX / VoIP Gateways through Open Source software is a viable alternative to the real needs of business communication.
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Depiak, Petr. "Bilingový systém a monitorování hovorů pro PBX Asterisk." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2010. http://www.nusl.cz/ntk/nusl-218271.

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This master's thesis is focused on developement of billing system with the options of monitoring individual calls for software exchange Asterisk. Billing of calls is adaptible with the help of group of individual rules, consisting of tariff impulses, numerical prefix, with help of outgoing trunk and cost of the billed unit. The first part of this work is focused on instalation, configuration and preparation of individual components of the billing system. In this work is explained the architecture of the billing system and highlighted the purpose of work of the model database. Next we focused on the purpose and the principal system invidual function of the system including solution. At last there is a simple manual to operate the system with the help of created web interface.
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Jakubíček, Michal. "Zvukový kodek s podporou zabezpečení pro PBX Asterisk." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2015. http://www.nusl.cz/ntk/nusl-220422.

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This thesis is focused on the design of secured audio codec for Asterisk PBX. The first chapter is focused on the basic division of traditional PBX producers and the open source PBX. The second chapter explains the structure of Asterisk PBX and its fundamental difference from a traditional PBX. Asterisk is based on components called modules, therefore the work also deals with the most important modules for operation of exchanges and their division of terms of support and dividing by the type of application and their properties. In this chapter there are described in more detail audio codec A-law and u-law. The third chapter contains simple instructions to get your orientation in the construction of the module for Asterisk PBX and this guide is accompanied by a simple example of creating a module demonstration of his method of translation, commissioning and loaded into Asterisk. Simulation of voice security is in the fourth chapter which provides a description of the proposed security solutions with subsequent implementation in Simulink. This simulation verifies the functionality of the solution proposed security phone call. In the fifth chapter outlines the historical use of encryption techniques primarily mirroring the spectrum and time division signal and comparing them with current modern digital technics. In the last sixth chapter is the actual implementation audio codec module with encryption.
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Janíček, Martin. "Možnosti implementace signalizačního systému číslo 7 v PBX Asterisk." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2009. http://www.nusl.cz/ntk/nusl-218068.

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Semestral project describes signaling system number 7, it's settings options and options of signaling over IP networks, especially two transport protocols SCTP and BICC for signaling SS7 over IP. Analyses kinds of implemetations of this signaling system to the Asterisk PBX with TDM E1 card support. Further part of this work is dedicated to the open source implementations libss7 of Digium and chan_ss7 which is currently developed by Dicea. Describes in detail their installation to the open source PBX Asterisk including testing of both and comparing these two open source solutions. Last part is focused on realization of gateway which converts communication from TDM network to IP network. For this part, three computers are used. First as SS7 signalling end softswitch, second as SIP signalling end softswitch and last as gateway between them. This gate works as interface between SS7 signalling and SIP signalling. Testing call was realized successfully for both directions.
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Šalko, Jaroslav. "Implementace WebRTC v Open source PBX." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2018. http://www.nusl.cz/ntk/nusl-377319.

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The topic of this work is verification of support WebRTC communication through selected Open Source PBX. This work examine demands for WebRTC communications and describes configuration of branch centers for this type of communication. In the theoretical part is reader acquainted with the term WebRTC and with protocols related to this kind of communications. The purpose of this part of the work is to bring the reader closer look to the principles of functioning to ensuring support for this kind of communications. This is also connected with Description of basic interfaces of WebRTC applications. Further the reader finds the configuration of the selected Open Source PBX so that they can make audio-video call between WebRTC clients. This section is divided into three subchapters, each of it deals with the same problems for one of the aforementioned PBX. At the end of each chapter where the PBX PBX is configured step-by-step, test calls are made. These calls are captured by the Wireshark packet analyzer and serve as a demonstration of the WebRTC configuration functionality. At the end of this section, PBXs are compared against each other about WebRTC support. Practical part is dealing with laboratory task for students which are studying subject telecommunication and information systems. In the task students will be configuring WebRTC for PBX Asterisk. The task contains brief description of WebRTC and comments for all steps for configuration. All steps and facts are demonstrated by exemplary configuration files.
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Binder, Tomáš. "Správa a konfigurace VoIP ústředny Asterisk." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217252.

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This diploma dissertation is dealing with the VoIP software exchange Asterisk. In the dissertation there are described its abilities and possible ways of its configuration. Special attention is given to the signalling protocol SIP, which is described in one of the chapters. Within this dissertation a dial plan, which demonstrates the technique of dial plan creating, was created. Within the boundaries of the dialplan following services could be found: a voicemail, conference, Interactive Voice Response and call queues. Configuration files, with the help of which the exchange is configurated, are described in my dissertation as well. Finally, three laboratory assignments for purposes of the subject Multimedia Services are mentioned. Their main aim is to familiarise students with the creation of SIP accounts in the exchange, their mutual connections, defining the Interactive Voice Response and forming a new call centre.
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Chalás, Jaroslav. "Metody zajištění bezpečnosti VoIP provozu Open source PBX." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2010. http://www.nusl.cz/ntk/nusl-218574.

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Main goal of creating the Open Source project and GPL licence are free sources and applications available for a wide public. Competent communities are responsible for support and upgrade of Open source based applications and softwares, which are created on a voluntary bases. Due to this fact an implementation depends on plenty others publicly available libraries and applications, which sometimes complicate the installation process itself. Successfully created VoIP connection is two-phase based process. Signalization is necessary in the first place, which might be supported with H.323 or SIP. After call parameter negotiation – voice codec, cipher code, ports etc, the second phase takes over to transfer voice. Theoretical part of this thesis describes SIP, H.323, MGCP, RTP and IAX protocols, as well as secure ways of signalization and voice stream part of the call. These might be SIPS, SRTP, ZRTP and IPsec. In thesis Open Source Asterisk PBX is well described, when mentioning its options, features and community support. I put near options available for particular releases and introduce attacks and abuses which are possible to perform on the VoIP system in general, together with available, no cost and working tools to perform the attacks with. Practical part focuses on possibilities to generate experimental attacks on individual systen parts with exact definition of what the consequences are. Based on the overall analyse of achieved results I conclude three solutions as autoinstallation linux packages. These „deb“ packages consist of specific Asterisk release required to meet the security needs, ready-to-test configuration and guide to follow with correct options to set. Final security possibilities requires hardening on application layer, where Iptables takes its part. „Linux firewall“ as some express Iptables are configured to reflect VoIP system parameters and protect from DoS attacks.
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Books on the topic "PBX Asterisk"

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Configuration Guide for Asterisk PBX. 2nd ed. BookSurge Publishing, 2007.

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Book chapters on the topic "PBX Asterisk"

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Xiang, Duo, and Li-hua Sun. "The Application of Asterisk-Based IP-PBX System in the Enterprise." In Lecture Notes in Electrical Engineering, 753–57. Berlin, Heidelberg: Springer Berlin Heidelberg, 2011. http://dx.doi.org/10.1007/978-3-642-21747-0_96.

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Gohel, Chirag K., and Kamaljit I. Lakhtaria. "Implement VoIP Based IP Telephony with Open Source Asterisk Architecture." In Research, Practice, and Educational Advancements in Telecommunications and Networking, 1–10. IGI Global, 2012. http://dx.doi.org/10.4018/978-1-4666-0050-8.ch001.

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Asterisk is a leading open source telephony software/system, easily implemented over intranet and internet. Asterisk empowers developers and integrators to create advanced communication solutions. An Asterisk system is a low cost type of a traditional PBX system. Any phone controlled by an Asterisk system can call a VoIP or analog phone controlled or managed by a traditional telephone system or by Asterisk telephone system. In this paper, the authors focus on the deployment and testing of various Open Source Asterisk Services in an enterprise level communication system. Selected services are listed in this paper that can be used to implement a telephone system with good Quality of Services (QoS) and good Quality of Experience (QoE) from the personal user to enterprise level users.
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Conference papers on the topic "PBX Asterisk"

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Costa, Lucas Rodrigues, Lucas Saad N. Nunes, Jacir Luiz Bordim, and Koji Nakano. "Asterisk PBX Capacity Evaluation." In 2015 IEEE International Parallel and Distributed Processing Symposium Workshop (IPDPSW). IEEE, 2015. http://dx.doi.org/10.1109/ipdpsw.2015.90.

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Ahmed, Mohiuddin, and Abdul Malik Mansor. "CPU dimensioning on performance of Asterisk VoIP PBX." In the 11th communications and networking simulation symposium. New York, New York, USA: ACM Press, 2008. http://dx.doi.org/10.1145/1400713.1400737.

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Goel, Saurabh, and Mahua Bhattacharya. "Speech based dialog query system over asterisk PBX server." In 2010 2nd International Conference on Signal Processing Systems (ICSPS). IEEE, 2010. http://dx.doi.org/10.1109/icsps.2010.5555468.

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Odjidja, Euclid, Salah Kabanda, William Agangiba, and Richard Annan. "WIRELESS VOIP IMPLEMENTATION USING ASTERISK PBX AND OPEN SOURCE SOFTPHONE." In EAI International Conference on Technology, R&D, Education and Economy for Africa. EAI, 2018. http://dx.doi.org/10.4108/eai.21-3-2018.2275638.

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Khan, Mohammad Azam, and Khaled Mahmud Shahriar. "ASTERISK Based Open Source IP-PBX System for Accountable Customer Support Service." In 2015 3rd International Symposium on Computational and Business Intelligence (ISCBI). IEEE, 2015. http://dx.doi.org/10.1109/iscbi.2015.22.

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Khan, Sarwar, and Nouman Sadiq. "Design and configuration of VoIP based PBX using asterisk server and OPNET platform." In 2017 International Electrical Engineering Congress (iEECON). IEEE, 2017. http://dx.doi.org/10.1109/ieecon.2017.8075808.

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Konshin, Sergey, M. Z. Yakubova, T. N. Nishanbayev, and O. A. Manankova. "Research and Development of an IP network model based on PBX Asterisk on the Opnet Modeler simulation package." In 2020 International Conference on Information Science and Communications Technologies (ICISCT). IEEE, 2020. http://dx.doi.org/10.1109/icisct50599.2020.9351405.

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Serikov, T. G., M. Z. Yakubova, A. D. Mekhtiev, V. V. Yugay, A. K. Muratova, V. P. Razinkin, A. V. Okhorzina, A. V. Yurchenko, and A. D. Alkina. "The analysis and modeling of efficiency of the developed telecommunication networks on the basis of IP PBX asterisk now." In 2016 11th International Forum on Strategic Technology (IFOST). IEEE, 2016. http://dx.doi.org/10.1109/ifost.2016.7884168.

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Santos, Johnny Cezar Marçal dos, Lindemberg Silva Pereira, and Sergio Vianna Fialho. "SAGA: Um Sistema WEB para Administração e Gerenciamento remoto e em tempo real de servidores VoIP baseados no Asterisk PBX." In XXVII Simpósio Brasileiro de Telecomunicações. Sociedade Brasileira de Telecomunicações, 2009. http://dx.doi.org/10.14209/sbrt.2009.55896.

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Cunha, Marcos, Karinne Silva, David Mota, and Alex Vasconcellos. "Uma arquitetura modular de hardware e software para PABX VoIP baseado em Asterisk." In XXX Simpósio Brasileiro de Telecomunicações. Sociedade Brasileira de Telecomunicações, 2012. http://dx.doi.org/10.14209/sbrt.2012.202.

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