Asterisk Research Papers - Academia.edu (original) (raw)

Create a Asterisk SIP server on your laptop an use two sip phone two make a call.

This study entails the simulation and implementation of voice over a data network, a telephony system using an IP PBX solution. A technology called voice over Internet Protocol (VoIP), or Internet telephony means that voice is carried... more

This study entails the simulation and implementation of voice over a data network, a telephony system using an IP PBX solution. A technology called voice over Internet Protocol (VoIP), or Internet telephony means that voice is carried over an IP network. Voice, which is an analogue signal, is converted to digital data, which is then disassembled and transmitted through a network only to be reconverted back to an analogue signal on the other end using a Linux based IP-PBX solution called Asterisk. This service can be properly managed and deployed over a network with less stress and expenses. The IP PBX main server also has integrated in it other communication services such as Voice mails, IVR’s, all embedded in the IP PBX SYSTEM. This technology promises an evolutionary leap beyond the standard telephone service we have been accustomed to, as well as a host of benefits. The new technology transmits voice signals the same way email is sent, using the Internet’s data-transfer protocols to break conversations into digital packets that can be sent on lower-cost, more efficient “packet-switched” networks. This project was able to address the persistent communication problem which existed in the departments by allowing users to communicate with the services the solution provided with less stress and comfort.
Target market includes: Corporate organizations, Universities, Health care, Airports, Hotels, Banks etc. This project is economic, cost effective, gives full control to the administrator and provides mobility, feasible, Peer-to-Peer phone calls. The contents of IP PBX System, supplemented by a good number of necessary and descriptive drawings which makes this project report very easy to understand

Resumen: Dentro de la categoría de los signos ortográficos, el asterisco es un signo auxiliar de la escritura, tanto manuscrita como impresa o electrónica. Su origen es contemporáneo de la creación de los signos de puntuación, aunque en... more

Resumen: Dentro de la categoría de los signos ortográficos, el asterisco es un signo auxiliar de la escritura, tanto manuscrita como impresa o electrónica. Su origen es contemporáneo de la creación de los signos de puntuación, aunque en la tradición gramatical latino-medieval aparece entre las notae sententiarum o notas críticas para el establecimiento correcto de los textos. Sus primeras definiciones reflejan el isomorfismo y las metáforas que asocian la claridad o presencia de luz a la inteligibilidad de las producciones verbales. Presentamos la historia del asterisco, así como un estudio comparado francés-español sobre su uso, junto con los del asterismo y el asterónimo.

SDN (Software Defined Networking) is a revolutionary technology that is currently indeed in demand in various network services. VoIP services can be greatly improved with SDN being used in their network. In this lab, we review the... more

SDN (Software Defined Networking) is a revolutionary technology
that is currently indeed in demand in various network services. VoIP
services can be greatly improved with SDN being used in their network.
In this lab, we review the technologies we are going to use in our
experiment and go through the background needed in our lab. Then,
we show how we built the lab itself in its different phases. Lastly, we
present the results we got from this lab. However, that doesn’t summarize
the capabilities of any of the used technologies but, it rather
gives a glimpse of what can be done with them especially SDN.

Elastix PBX Default Login Details and Default Passwords

Tahun Baru 2015 Masehi di bulan januari ini kita mulai mendalami tentang VoIP biar bisa telponan hemat biaya bahkan bisa di bilang bisa gratis. Kali ini kita coba bahas tentang cara menggunakan VoIP Client contohnya ZoIPer, cek aja di... more

Tahun Baru 2015 Masehi di bulan januari ini kita mulai mendalami tentang VoIP biar bisa telponan hemat biaya bahkan bisa di bilang bisa gratis. Kali ini kita coba bahas tentang cara menggunakan VoIP Client contohnya ZoIPer, cek aja di website nya www.zoiper.com

VoIP (Voice over Internet) provides delivery of voice information over unsecured IP-based networks like the Internet. VoIP data, signaling and voice, needs to be secured in such an environment. Security mechanisms take their toll on VoIP... more

VoIP (Voice over Internet) provides delivery of voice information over unsecured IP-based networks like the Internet. VoIP data, signaling and voice, needs to be secured in such an environment. Security mechanisms take their toll on VoIP system performance. SIP is dominant signaling protocol for VoIP. This paper measures relative decrease in VoIP performance of system with secured SIP signaling over one without it. It compares SIP with authentication enabled over three transport protocols: UDP, TCP and TLS. Peak throughput of concurrent calls, registration request delay, session request delay, SIP server CPU and RAM usage are measured. Testbed environment consists of Asterisk IP private branch exchange (PBX) as a part of Elastix server, several SIP user agents and SIPp traffic generator. Test results show that performance of SIP over TLS based signaling is four times lower than the SIP signaling over UDP in most metrics.

Telecommunications in recent years has been undergoing a rapid growth all over the world. Prominent among the advancements of telecommunication is the evolution of modern converged network that provides the trio of data, voice and video... more

Telecommunications in recent years has been undergoing a rapid growth all over the world. Prominent among the advancements of telecommunication is the evolution of modern converged network that provides the trio of data, voice and video network on a single network platform. The converged technology uses the internet as a medium to transmit data, voice and video packets using packet switching. This comes with numerous benefits by providing add-ons to both the service providers and users. The VoIP technology has been widely accepted and has received a boost in most western countries like the US and UK where users are now migrating from the legendry Public Switch Telephony Network (PSTN) to VoIP because of the numerous benefits it offers but unfortunately the business environment in Nigeria is yet to key into this promising technology. This study therefore focuses on introducing and implementing this technology in a converged network in Nigerian environment. It will also showcase VoIP’s numerous advantages and look at issues likely to be encountered during its implementation. It is hoped that it will help to serve as a tool in decision making process by converging the VoIP network with the already available data network.

Ce mémoire a été rédigé à la suite d'un stage académique effectué du 01 Juillet 2017 au 30 Septembre 2017 au Ministère de la Femme, de la Solidarité Nationale et de la Famille. Ce stage effectué à la fin de notre formation en cycle de... more

Ce mémoire a été rédigé à la suite d'un stage académique effectué du 01 Juillet 2017 au 30 Septembre 2017 au Ministère de la Femme, de la Solidarité Nationale et de la Famille. Ce stage effectué à la fin de notre formation en cycle de Licence d’Ingénierie des Systèmes et Réseaux à Sup’Management Burkina, nous a permis non seulement de nous imprégner des réalités du milieu professionnel qu'est l'entreprise mais aussi et surtout de faire l'adéquation entre la pratique et la théorie issue des enseignements reçus en matière de systèmes et réseaux.
La VOIP est une technologie qui s'impose progressivement dans tous les secteurs, elle consiste à faire transiter les communications téléphoniques par le réseau IP. Elle est aujourd'hui de plus en plus déployée au sein des grandes entreprises. Le développement de la VoIP est parti d'un simple constat : comment faire en sorte d'utiliser les potentialités du réseau IP afin de téléphoner moins cher (voire gratuitement) ? C'est dans le but de convergence sur les réseaux qu'est apparu le protocole SIP issu des grands opérateurs réseaux. IL est souple et évolutif.

This paper is concerned with the appropriation of the affordances of text-based communication in digital media to evoke associations with multimodal communication, specifically visual, auditory and haptic experiences accompanying observed... more

This paper is concerned with the appropriation of the affordances of text-based communication in digital media to evoke associations with multimodal communication, specifically visual, auditory and haptic experiences accompanying observed nonverbal phenomena and actions in text-messages. In order to account for these phenomena, the notion of kineticon is introduced and theorised from the perspective of its constitutive elements, established conventions, and functions. Through the analysis presented here, I identify a user-initiated language development serving to express multimodal meanings within a written medium often simplistically treated as mono-modal. I also demonstrate that the Goffmanian categories of given and given off expression need to be reconsidered in the light of the emergence of the expression of multimodal content in text-based digital media. The paper proposes a methodological approach to the analysis of user-initiated language phenomena, which includes naturally occurring data collection, the use of online participant observations, and detailed interviews using data as prompts.

Implementations of Voice over Internet Protocol (VoIP) have focused, up to now, mainly on the need to transport data in real-time, often at the expense of security. The neglect of secure VoIP is often intentional, as developers are... more

Implementations of Voice over Internet Protocol (VoIP) have focused, up to now, mainly on the need to transport data in real-time, often at the expense of security. The neglect of secure VoIP is often intentional, as developers are striving to minimise overheads and delays. The Secure Real-Time Protocol (SRTP) has the potential to secure real-time streams without exacting too high a performance price. SRTP is the addition of security to the audio/video profile used in the Real-Time Transport Protocol (RTP). SRTP adds confidentiality, integrity and optionaly authenticity to RTP media streams. This paper focuses on the integration of SRTP into Asterisk, an open-source VoIP PBX. SRTP support has recently been added to Asterisk by Mikael Magnusson. This paper analyses Magnusson’s implementation, contrasting it to a proof-of-concept implementation developed independently at Rhodes University. The interoperability of SRTP implementations cannot be taken for granted, given the relatively r...

We are today living in age where technology continues to bring about one magical transformation after another into our lives. Especially computers and internet that constitute the cardinal components of the present day technology have so... more

We are today living in age where technology continues to bring about one magical transformation after another into our lives. Especially computers and internet that constitute the cardinal components of the present day technology have so penetrated into our live that they today make up an indispensable part of life. In this paper, we present an Asterisk based framework that is designed particularly to the benefit of the visually challenged people. These people find it difficult to use the conventional computer access devices like the keyboard, the mouse or the monitor as these devices require a good hand-eye coordination. Our framework is designed to take DTMF input from the user that serve as commands for execution of different operations on the computer and provide output in the form of voice using speech synthesis. Additionally, our framework provides the benefit of customizing it as per user needs as it is based on the open source platform Asterisk.

In this paper, we studied the Voice over IP Quality of Service measurement techniques and Quality of Service implementations. We find out techniques that gives better Quality as well as their weaknesses. We have studied E-Model, Mean... more

In this paper, we studied the Voice over IP Quality of Service measurement techniques and Quality of Service implementations. We find out techniques that gives better Quality as well as their weaknesses. We have studied E-Model, Mean Opinion Score (MOS) optimization and other traditional algorithms as well as soft computing methods to find Quality of Service of Voice over IP. A thorough study of getting maximum Quality of service is presented here.

In particolare il lavoro riguarda la realizzazione di un servizio di conferenza telefonica/VoIP multiutente di cui ho curato l'analisi delle funzionalità da fornire all'utente, l'analisi delle problematiche di scalabilità e affidabilità e... more

In particolare il lavoro riguarda la realizzazione di un servizio di conferenza telefonica/VoIP multiutente di cui ho curato l'analisi delle funzionalità da fornire all'utente, l'analisi delle problematiche di scalabilità e affidabilità e l'implementazione di un prototipo del prodotto finale.

... Mosiuoa Tsietsi, Zelalem Shibeshi, Alfredo Terzoli, George Wells Rhodes University Department of Computer Science Grahamstown, South Africa ... Additionally, web-based systems are considered to be the best type of system to address... more

... Mosiuoa Tsietsi, Zelalem Shibeshi, Alfredo Terzoli, George Wells Rhodes University Department of Computer Science Grahamstown, South Africa ... Additionally, web-based systems are considered to be the best type of system to address the needs of a live virtual classroom [2]. ...