Dissertations / Theses: '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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Stračár, Ivan. "Implementace jednoduché pobočkové ústředny na OpenWRT." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2014. http://www.nusl.cz/ntk/nusl-220608.

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The diploma thesis deals with the system OpenWRT. Installing this system on the router Siemens Gigaset SX762. Describes how to compile and upload the simple package helloworld into this system. The package was tasked invitation simple phrase "Hello World" to the system console of OpenWRT. Package only serve to show that the system OpenWRT allows users to customize it according to their needs. After that it was installed PBX Asterisk into the system OpenWRT. Proper functioning of PBX Asterisk has been verified to make a call between two software phones ZoiPer. Furthermore, the work described telephony application programming interface (TAPI). Some of its fun- ctions, interfaces and packages needed to communicate with the system OpenWRT. In conclusion, the presented test topology and verify the operation of making calls between analog as well as softphones.

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Bednár, Jakub. "Výkonnostní limity, spolehlivost a bezpečnost Open source PBX." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2014. http://www.nusl.cz/ntk/nusl-220951.

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The aim of this thesis is to install and to configure three Open source PBXes Asterisk, Freeswitch and YATE. Furthermore, the aim is to realize the performance test and stability tests on three different HW configurations with the tester Spirent Abacus 5000. The scripts in bash were created to monitor PBX performance. Another part of the study is to analyze and to compare PBX security and to compare the Open Source PBX with a proprietary PBX Alcatel-Lucent OXE.

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Melichar, Ondřej. "Pobočková VoIP ústředna Asterisk a její nástavby." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2018. http://www.nusl.cz/ntk/nusl-377036.

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This master’s thesis delves into the possibilities of the open-source Private Branch Exchange Asterisk, elaborates on its features and compares it with several other distros. The term SIP stack is explained here with the mention of two of its representatives. Further in the thesis, the security risks of the VoIP technology are explained, and specific attacks are described and then realized. As a part of the testing process, the possibilities of a custom module and its following implementation are explored, as well as the portability between the individual distros and its proper functioning.

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Papež, Nikola. "Implementace protokolu SIP v open Source PBX a jejich testování." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2016. http://www.nusl.cz/ntk/nusl-242146.

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This diploma thesis examines and compares several selected libraries of SIP protocol, performance, stability, security and impact of their configuration. The main functions of the signalling protocol are briefly named at the beginning. The following chapters describe the tested PBXs and several stacks for SIP protocol are theoretically compared. The practical part deals with measurements conducted on the load generator Spirent TestCenter C1 which is used for all the performed tests on exchanges. All the mentioned SIP libraries, PBXs and the operating system on which the PBXs were running are open source software.

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Kakvic, Martin. "Metody zabezpečení IP PBX proti útokům a testování odolnosti." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2015. http://www.nusl.cz/ntk/nusl-220331.

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This diploma thesis focuses on attacks on PBX Asterisk, FreeSWITCH and Yate in LTS versions. In this work was carried out two types of attacks, including an attack DoS and the attack Teardown. These attacks were carried out using two different protocols, SIP and IAX. During the denial of service attack was monitored CPU usage and detected if its possible to establish call and whether if call can be processed. The Security of PBX was build on two levels. As a first level of security there was used linux based firewall netfilter. The second level of security was ensured with protocols TLS and SRTP.

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Sandström, Kristoffer. "Migrering till IP-baserad telefonilösning." Thesis, KTH, Skolan för informations- och kommunikationsteknik (ICT), 2015. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-175390.

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Användandet av IP-telefoni har de senaste åren ökat och marknaden förutspås fortsätta att växa globalt. Många vill ta del av de fördelar som den nya tekniken har i form av ny funktionalitet och minskade kostnader. Men att migrera telefoni till datanät medför både nya möjligheter men också nya utmaningar. I den här rapporten undersöks hur Asterisk kan användas som ett bra IP-PBX alternativ. Rapporten behandlar även säkerheten i att ansluta ett system med Asterisk till internet genom intrusionstester som utförs på systemet i grundkonfiguration. Dessa tester resulterar i rekommendationer om hur systemet kan konfigureras för att hålla en hög säkerhetsnivå.
The usage of IP-telephony has increased in recent past and the market is expected to continue to grow globally. Many want to take part in the advantages that the new technology brings in form of functionality and reduced costs. But to migrate telephony to data networks brings both new possibilities but also new challenges. This report examines how Asterisk can be used as a good IP-PBX alternative. The report also addresses the security aspect of connecting a system based on Asterisk to the internet through conducting intrusion tests on the system in standard configuration. These tests result in recommendations on how the system can be configured to keep a high security standard.

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Hynek, Luboš. "Metody zajištění IP PBX proti útokům." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2013. http://www.nusl.cz/ntk/nusl-220328.

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This master project focuses on the possibilities of protecting the most common free software PBX Asterisk, FreeSWITCH and YATE. In practice, it was verified the behavior of PBX in the attacks and suggested protection against them on one of the most popular distributions of Linux server on CentOS. Tool was created to simulate several types of attacks targeting denial of service. Both protective options PBX themselves and operating system capabilities are used in this work. Comparison was also the possibility of protection of individual PBX with each other. It also includes a brief description of the protocol, topology attacks and recommendation for the operation of softswitches.

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Bílek, Petr. "Měření kvality telefonních hovorů u pobočkové ústředny Asterisk." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2011. http://www.nusl.cz/ntk/nusl-218970.

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This thesis contains description of the present methods and algoritms for measuring the quality of telephone call on the PBX Asterisk. Further develop the concept of a system for measuring and implementing the call embbeded system VOIPAC PXA270M. Based on theoretical analysis is carried out discussions between C, Java and PHP,which results in the selection of appropriate programming language suitable for implementation in embedded system. The concept is realized by creating an application written in a particular programming language. Part of this work is verification created application using experimental measurments of telecommunications system. The results of experimental measurements are discussed in the final evaluation.

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Božek, Martin. "Open IMS Core a IP Multimedia Subsystem." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2011. http://www.nusl.cz/ntk/nusl-218969.

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This thesis describes architecture of IMS and shows possibilities of IMS platform testing. Theoretical part describes layer model of the IMS as a whole and then describes it’s individual layers. Next chapters analyse key entities of the IMS, interconnection between reference points and features of protocols used in the IMS. Practical part deals with the introduction of Open IMS Core, which was chosen for the IMS technology testing. Settings necessary to carry out testing and interconnection between PBX Asterisk are shown in next chapters. After introduction of IMS desktop clients is carried out an instant messaging communication within the IMS network. The communication is captured and analysed by Wireshark application. Afterwards there is described how SIP protocol sends messages within the IMS. After a brief introduction to the PBX Asterisk, there are discussed assumptions for the interconnection between Asterisk and IMS. There are also described necessary settings needed for implementation and communication testing itself. The first test is an audio session carried out between the desktop IMS client and IP phone registered to the PBX Asterisk. Communication is captured for the analysis of preparation, conduction and termination of the session. After the successful realization of the audio call, video session has been made. The session was analysed in detail, including statistics of control signals and transmitted packets. There are two laboratory excercises in attachement of this thesis, which will help students to understand the IMS technology and communication options within the IMS network

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Al-Anqari, Mhannad. "Možnosti přenosu signalizace SS7 přes IP síť s využitím ústředny YATE." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2013. http://www.nusl.cz/ntk/nusl-219912.

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This study examines the use of SS7 signaling system over IP networks by using the open source PBX YATE. At first it starts with describing the SS7 followed by an explanation of the function of each of its levels and the messages that are used within the SS7 network. The study then sheds some light on the ways of using SS7 inside IP network with the use of some protocols. It also discusses the architecture of YATE and its files, and how it is installed in Linux operating system. Finally, it describes the important files for delivering this task. The study was commenced by using two virtual machines that have two different open source PBX's which are YATE and Asterisk, and after acquiring some results by establishing communication between them via the means of SIP trunk, furthermore the study was extended to the laboratory in order to test it over real servers that have TDM cards, in order to apply the study by the means of SS7 protocols, SIGTRAN, MGCP gateway and SIP-T. The experiments have almost delivered successful communications after conducting a configuration for the files on multiple sides.

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Weltz, Max. "Dial over Data solution." Thesis, KTH, Kommunikationssystem, CoS, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91874.

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The increased use of computer networks has lead to the adoption of Internet-based solutions for reducing telephony costs. This has proved to be a boon to callers who can reach the other party directly via the Internet. Unfortunately numerous business persons still need to call to and from mobile phones which are currently a domain where the customers are generally tightly bound to their operators. To provide a simple solution to this problem for companies, Opticall AB has designed an integrated system called the Dial over Data solution, coupling a mobile interface with a low-rate communication channel, which allows calls to be originated remotely at the best price, exploiting the customer company's existing network. This scheme allows the customer company to easily control telecommunications costs, to monitor their employees' efficiency, and more generally speaking to claim a central role in the communications of their employees. The proposed solution allows distant callers (usually employees of the customer company) to benefit from the company's internal network, which is usually more cost effective and offering connectivity to more networks than a cell phone. The Dial over Data solution enables communication between any phone accessible from the customer company's telephony network (such as SIP clients, landline phones, and mobile phones) at a lower cost.</p> This thesis project analyzes existing technologies and compares them to the pre-existing prototype to ascertain the validity of the method and of the components used. This project also explains the improvements brought to the features offered by the DoD solution: the initial prototype has been developed into a stable and functional product, and has been tested internally. Prompted by a need for scalability and additional features, the replacement of Asterisk for the handling of SIP calls by other SIP servers has also been considered and tested.
Den nuvarande ökningen av datanätverk har lett till adoptionen av Internetbaserade lösningar för att förminskar kostnader inom telefoni. Tyvärr behöver åtskilliga affärsmän fortfarande ringa till och ifrån mobiler som återstår som ett område där kunderna är fastkedjade till deras operatörer. För att tillföra en enkel lösning till detta problem för kundföretag har Opticall AB planlagt ett integrerat system som kallas Dial over Data som kopplar ihop ett mobilt gränssnitt, med en billig kommunikationsmedel, som tillåter telefonsamtal påbörjas avlägset på det billigaste priset tack vare företagets nätverk. Det ger möjligheten till kundföretaget att vara centralt för sina personals kommunikationer. Det medger ett enkelt sätt att kontrollera kostnader samt övervaka personalens effektivitet. Den Dial over Data lösningen är en lösning som tillåter avlägsen besökarna med kundföretagets inre nätverk kommer att dra nytta av eftersom det är mer kostnadseffektiv och flexibel än en blott mobiltelefon. Denna möjliggör kommunikation mellan SIP-klienter, fast telefoni och mobiltelefoni för en lägre kostnad till företaget utan att framkalla besvär för sina anställda. Konnektiviteten till företagets inre nätverk samt en låg besvärlighetsnivå är garanterade respektive genom konfigurationsförmågan av produkten och ett praktiskt gränssnitt som är redo för korporativkontaktlistor och visar alla informationen som är relevanta till förbrukarnas erfarenhet.</p> Det här avhandlingsprojektet analyserar existerande teknologier och sätter dem i relation till den sedan tidigare framtagna prototypen för att utröna validiteten hos metoden och beståndsdelar. Projektet förklarar även de förbättringar som gjorts till de egenskaper som erbjuds av DOD-lösningen: prototypen har utvecklats till en stabil och funktionell produkt och har testats internt. Driven av behovet för skalabilitet och ytterligare egenskaper har ersättandet av Asterisk för hantering av SIP samtal av andra SIP servers övervägts och testats.

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Zelenay, Martin. "Testování odolnosti IP PBX proti útokům s využitím testeru Spirent Avlanache." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2015. http://www.nusl.cz/ntk/nusl-220404.

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This work explores, analyzes and rate infuence of VoIP attacks on open source pbx functionality. It describes how voice over IP attacks are achieved according to security standards. There are described concepts and basics of VoIP networks with orientation on facts necessary to understand analyzed actions and measurements in theoretical part. In practical part, there is described realization of attack tests according to instructions with first orientation on initial checking of testing device, pbx’s and security attack possibilities and then complex creation and testing of attack scenarios types such as fuzzing and denial of service attacks.

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Halamík, Zdeněk. "Možnosti vazby softswitche Asterisk na pobočkové ústředny 4. generace." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217329.

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This master’s thesis dissertate the possibilities of the linkage between Asterisk softswitch and the 4th generation private branch exchange. This should create a new generation’s network, so-called NGN, by the convergence of existing telecommunication networks with an IP computer network. This master’s thesis is divided into several chapters. In introduction is described the evolution of the private branch exchanges as well as the principles of the voice digitizing, codecs and signaling commonly used in both TDM and VoIP networks. The main aim of this project is the configuration of Asterisk software exchange for connection with PBX Alcatel 4400 as well as public phone network PSTN. Another goal of this master’s thesis was the configuration of Alcatel PBX and diagnostics of CCS and CAS signaling on E1 interface. In conclusion there are summarized advantages of NGN networks and their utilization in the future.

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Konečný, Jakub. "Návrh a implementace interaktivního grafického rozhraní pro IVR." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2020. http://www.nusl.cz/ntk/nusl-413192.

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This diploma thesis focuses on development of graphical user interface for managing IVR applications. The work is more software oriented, it analyzes current state of the iPBX product belonging to the IPEX a.s. company, describes used technologies and introduces new concept of interactive user interface for generating IVR diagrams together with Asterisk dial plan generator.

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25

Bergsten, Jonas. "ATT VÄLJA RÄTT IP-TELEFONILÖSNING : En jämförande studie av mjukvara." Thesis, Högskolan i Skövde, Institutionen för kommunikation och information, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:his:diva-5995.

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I detta arbete har vanligt förekommande krav på IP-telefonilösningar identifierats via litteraturstudier. Därefter har en testmiljö skapats där ett urval av mjukvaror har utvärderats med avseende på funktionalitet och dokumentation. Kraven som ställts relaterar till säkerhet och kvalitet samt den funktionalitet som krävs för att motsvara den hos det vanliga telefonnätet. Mjukvarorna som presenteras är 3CX, Asterisk, sipXecs och Switchvox. Urvalet av mjukvaror är baserade på antingen öppen källkod eller proprietära motsvarigheter. Faktorer som möjlighet till support och användarvänlighet har också belysts. De mjukvaror som presterar bäst utifrån givna parametrar är 3CX och Asterisk. Arkitekturen mellan de två mjukvarorna olika och båda kan ha fördelar beroende på struktur och kompetens hos det specifika företaget.

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Starzyczny, Radek. "Multiplatformní brána pro hlasovou komunikaci v reálném čase." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2015. http://www.nusl.cz/ntk/nusl-220415.

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This master's thesis is focused on VoIP communications. It describes deploy of the operating system OpenWRT, analog interface of router Gigaset SX762 and GSM gateway for receiving or place calls. The paper describes the protocols involved in the communication and basic configuration elements. Deploying IP telephony enables to reduce the cost of operation and provides a number of additional functions.

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27

Brejcha, Martin. "Vývoj ovladače pro zákaznický analogový uživatelský modul v OS Linux." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2009. http://www.nusl.cz/ntk/nusl-218140.

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This master's thesis describes how to develop loable kernel module for operating system Linux. Module can be use like driver for concrete hardware device. In this case for telecommunication hardware. The second part of this thesis describes how to implement support for this hardware in Asterisk PBX. Support in Asterisk is realized by channel module. In that channel module are implemented functions for process incoming and dialed calls.

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Lindecrantz, Mikael, and Marcus Junström. "VoIP som kommunikationsplattform : - Tjänster och möjligheter." Thesis, Linköping University, Department of Management and Engineering, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-9700.

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Begreppet Voice over Internet Protocol (VoIP) är något som nog de flesta har hört talas om, men vad innebär egentligen VoIP. Vi har tittat på tekniken och återger en beskrivning av begreppet VoIP. Vi har valt att titta på hur företag använder tekniken samtidigt som vi tittar på vilka nya tjänster och möjligheter som finns. Hur ser tjänsteutbudet ut idag och hur påverkar detta sättet att kommunicera inom företagen? Vilken funktionalitet används och hur drar man nytta av de fördelar och möjligheter som den nya tekniken erbjuder? Med förändrade kommunikationssätt ser vi hur verksamheter både kan effektiviseras och utvecklas.

När vi tittar på hur leverantörer bidrar till att utveckla sina kunders verksamheter så ser vi att leverantörerna inte är delaktiga inom detta område i någon större utsträckning. Vi ser dock att man inte hunnit så långt i utvecklingen av VoIP, samt att leverantörer har vissa problem med att nå ut och marknadsföra de tjänster och möjligheter den nya tekniken faktiskt erbjuder. Fokus ligger istället mer på de kostnadsbesparingar man gör genom att kommunicera över internet (IP) istället för med traditionell telefoni (PSTN).

Tekniken är beroende av kvaliteten på bredbandsuppkoppling samt den interna infrastrukturen hos företagen. Mycket av de problem så som eko och dålig samtalskvalitet beror på undermålig utrustning internt på företagen. Dessa problem ser man i branschen som något övergående då ny hårdvara utvecklas och förses med bättre stöd för VoIP. Många leverantörer erbjuder även sina kunder en helhetslösning för att få kontroll över både telefoni och internetförbindelse.

Slutligen ser vi att VoIP öppnar upp för en omstöpning av hur företag kommunicerar, möjligheterna har bara börjat utforskas. Som teknik är VoIP idag inget nytt, det som är nytt är att företagen först nu börjar se möjligheterna till att effektivisera, utveckla och förändra sina kommunikationsvägar. För att utfallet ska bli så bra som möjligt krävs ett helhetsgrepp på hur VoIP implementeras, vi menar att man måste se till hur man kan förändra sitt arbetssätt, inte bara hur man ringer till en lägre kostnad.

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Abreu, Marcelo Pereira de. "Implantacão de um sistema de telefonia IP em uma rede sem fio: VoIP Móvel." Niterói, 2017. https://app.uff.br/riuff/handle/1/3933.

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Submitted by Patrícia Cerveira (pcerveira1@gmail.com) on 2017-06-13T15:41:09Z No. of bitstreams: 1 Marcelo Dissertação.pdf: 13477166 bytes, checksum: e46f91138593316f4c2c1504aaf297a3 (MD5)
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O serviço ”Móvel de Voz sobre IP” é a convergência natural da tecnologia de voz sobre IP (VoIP) e a comunicação sem fio, e pode impulsionar o aumento da popularidade da primeira bem como promover constantes avanços da última. Embora seja possível encontrar várias aplicações que oferecem serviço de VoIP na Internet e muitos dispositivos que implementam VoIP em hardware, uma implementação aberta e não-proprietária pode ser integrada aos serviços legados - como PABXs institucionais - o que proporciona uma contribuição significativa. Este trabalho descreve a implementação do serviço Móvel de Voz sobre IP no Instituto Federal Fluminense e destaca os desafios a serem enfrentados em seu gerenciamento e operação, enfatizando a segurança contra ataques. Os principais resultados indicam que este serviço oferece flexibilidade, conforto, redução de custos e mobilidade para o serviço de voz.
A “Mobile Voice over IP” service is the natural convergence of Voice over IP (VoIP) technology and wireless communication, and can leverage the increading popularity of the former and the constant advances of the latter. Although we can find various applications that o er VoIP service on the Internet, and in e ect many devices that implement VoIP in hardware, an open, non-propietary implementation that can be integrated with legacy services such as institutional PABXs is a welcome addition. This works describes the implementation of the Mobile Voice over IP service in the Instituto Federal Fluminense, and the challenges of its management and operation, with emphasis in security against attacks. This services brings flexibility, confort, cost reduction and mobility to the voice service.

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Dissertations / Theses on the topic 'PBX Asterisk'

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