A Proposal for a High Availability Architecture for Voip Telephone Systems Based on Open Source Software
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Skype for Asterisk™ Administrator Manual
Skype for Asterisk™ Administrator Manual 601-00017 Rev. B2 Digium, Inc. 445 Jan Davis Drive NW Huntsville, AL 35806 United States Main Number: 1.256.428.6000 Tech Support: 1.256.428.6161 U.S. Toll Free: 1.877.344.4861 Sales: 1.256.428.6262 www.asterisk.org www.digium.com www.asterisknow.org © Digium®, Inc. 2010 All rights reserved. No part of this publication may be copied, distributed, transmitted, transcribed, stored in a retrieval system, or translated into any human or computer language without the prior written permission of Digium, Inc. Digium, Inc. has made every effort to ensure that the instructions contained in this document are adequate and error free. The manufacturer will, if necessary, explain issues that may not be covered by this documentation. The manufacturer’s liability for any errors in the documents is limited to the correction of errors and the aforementioned advisory services. This document has been prepared for use by professional and properly trained personnel, and the customer assumes full responsibility when using it. Adobe and Acrobat are registered trademarks, and Acrobat Reader is a trademark of Adobe Systems Incorporated. Asterisk, Digium, Switchvox, and AsteriskNOW are registered trademarks and Asterisk Business Edition, AsteriskGUI, and Asterisk Appliance are trademarks of Digium, Inc. Any other trademarks mentioned in the document are the property of their respective owners. Digium, Inc. Page 2 TABLE OF CONTENTS Chapter 1: Overview.................................................................................................................6 -
Sangoma T116 16-Span T1/E1/J1 Tapping Board
Sangoma T116 16-Span T1/E1/J1 Tapping Board Dedicated tapping solution for up to 8 two-way connections or 16 one-way connections. The T116 Tapping Card is part of Sangoma’s family of Advanced Flexible Telecommunications hardware product line — it uses the same high-performance PCI Express interface that is providing superior performance in critical systems all over the world. The T116 supports the passive tapping of up to 240 voice calls using up to 16 T1, E1 or J1 spans. With Sangoma cards, you can take advantage of hardware and software improvements, as soon as they become available. The T116, like all cards in Sangoma’s AFT family, is eld upgradable with unbreakable rmware. Choose the T116 to collect call control information, telecom protocol information and voice/media. T116 Card Features Sixteen receive-only spans with Supports Robbed Bit Channel optimum PCI-Express interface Associated Signaling (CAS) and ISDN enables tapping of sixteen one-way PRI or eight two-way conversation T1/E1 and fractional T1/E1, multiple Support for Asterisk®, Yate®, and channel HDLC per line for mixed 5 Year warranty on parts and labor FreeSWITCH® PBX/IVR Projects, as data/TDM voice applications well as other open source and Supports the passive tapping of up proprietary PBX, Switch, IVR, or VoIP WANPIPE® routing stack is to 240 voice calls using up to 16 gateway applications completely independent of TDM T1, E1 or J1 spans voice application for total system Optimized per channel DMA streams reliability and hardware-level HDLC handling Field -
Voip Tutorial
VoIP and FreeBSD The daemon meets the phone May 15th, 2008 University of Ottawa, , Ottawa, Canada Massimiliano Stucchi [email protected] May 16th, 2007, Agenda • Introduction • Terms • Introduction to Asterisk key concepts • Let's connect to a provider • What's a dialplan ? • How cool is an IVR... • (if time permits) AGI overview 2 Who am I ? • First of all, good morning • I’m Max, nice to meet you all • I worked on VoIP and FreeBSD for the last 3,5 years implementing technologies for businesses large and small • I’m now working at BrianTel Srl, delivering voice services over geographical Wi-Fi Networks (more on this @ other talk) Why am I doing this ? • VoIP is business, but you have to know how to deal with it (and have right equipment) • It's not rocket science, but it's a totally different environment for computer professionals • I have some deal of experience • I don't feel the need to keep others from doing what I do 1000's miles away • It's fun ! Let’s start • A few questions to let me understand the level of knowledge of the class • If you have any question further on, raise your hand at any time any time TERMS Terms 1/5 • Direct Inward Dial • It's a real PSTN number which lets the call into your VoIP system. • Normally works on a PRI DID • It's normally intended as a phone number • Can be bought from many different providers and forwarded to your asterisk box via any provider Terms 2/5 • Voice Circuit (may carry data as well) • Can carry either 24 (T1) or 30 (E1) b- channels (audio) and 1 d-channel (for data communication across peers). -
Asterisk Record Codec
Asterisk record codec Record(filename:format[|silence][|maxduration][|option]) video portion of the recording is automatically set to the active video codec (Asterisk. Asterisk CodecsAsterisk supports the following narrow-band and kHz wideband codec; passthrough, playback and recording in Asterisk ;. ord A Call NativelyDescriptionMixMonitor. Current Documentation Record A Call Natively. Asterisk has issue regarding video codec negotiation; Advanced When you record a message to a voicemail, Asterisk records video too. As a part of the Media Overhaul project for Asterisk 10, changes have been made to Asterisk to increase the number of codecs it's capable of. Hi all, i'm trying to record a video call between two SIP client using ast_translator_build_path: No translator path: (ending codec is not valid). However if you playback something and recording is not in.g you need codec. If you use uncompressed stream(other codec or pstn/e1. Digium's implementation of the G codec allows Asterisk software to of Asterisk such as Call Conferences, DTMF digit collection, Call Recording and more. Asterisk is going to expect that all audio conforms to this standard. There are a lot of codecs that are used to compress the audio, the most common being ULAW. Also, this paves the way for other codecs under the G umbrella. Added support for recording of Asterisk voice calls (TDM and IP) using Xorcoms Asterisk. In the early days I used asterisk to record the prompts. It's not bad but you're limited by the quality of your phone and the codec you use. Asterisk can forward video with compressed H, but it can't act as a gateway to The standard video codecs can play recorded movies, clips, or other. -
Guide to Open Source Solutions
White paper ___________________________ Guide to open source solutions “Guide to open source by Smile ” Page 2 PREAMBLE SMILE Smile is a company of engineers specialising in the implementing of open source solutions OM and the integrating of systems relying on open source. Smile is member of APRIL, the C . association for the promotion and defence of free software, Alliance Libre, PLOSS, and PLOSS RA, which are regional cluster associations of free software companies. OSS Smile has 600 throughout the World which makes it the largest company in Europe - specialising in open source. Since approximately 2000, Smile has been actively supervising developments in technology which enables it to discover the most promising open source products, to qualify and assess them so as to offer its clients the most accomplished, robust and sustainable products. SMILE . This approach has led to a range of white papers covering various fields of application: Content management (2004), portals (2005), business intelligence (2006), PHP frameworks (2007), virtualisation (2007), and electronic document management (2008), as well as PGIs/ERPs (2008). Among the works published in 2009, we would also cite “open source VPN’s”, “Firewall open source flow control”, and “Middleware”, within the framework of the WWW “System and Infrastructure” collection. Each of these works presents a selection of best open source solutions for the domain in question, their respective qualities as well as operational feedback. As open source solutions continue to acquire new domains, Smile will be there to help its clients benefit from these in a risk-free way. Smile is present in the European IT landscape as the integration architect of choice to support the largest companies in the adoption of the best open source solutions. -
Analysis and Testing of Voip- Subsystems of Ipbrick
ANALYSIS AND TESTING OF VOIP- SUBSYSTEMS OF IPBRICK SWAMY H K Julho de 2017 1 ANALYSIS AND TESTING OF VOIP-SUBSYSTEMS OF IPBRICK PORTUGAL, S.A. SWAMY HK Department of Electrical Engineering Master‟s in Electrical Engineering – Power Systems 2017 2 3 Report prepared for the partial satisfaction of the requirements of DSEE Unit Course – Dissertation of the Master‟s in Electrical Engineering – Power Systems Swamy HK, Nº 1150084, [email protected] Professor.Sergio Ramos , [email protected] Company: IP BRICK, Oporto, Portugal Supervisor: Mr. Miguel Ramalhão, [email protected] Department of Electrical Engineering Master‟s in Electrical Engineering – Power Systems 2017 4 5 Where there is righteousness in the heart, There is beauty in the character. When there is beauty in the character, There is harmony in the home. When there is harmony in the home, There is order in the nation. When there is order in the nation, There is peace in the world.” - APJ Abdul Kalam 6 7 ACKNOWLEDGMENT At the end, the last stage of many successful stages over the course of my academic career comes to an end and its pride and pleasure that I completed it. During the development of this thesis work, the support and encouragement of several people were fundamental. So I would like to thank all those who have left their footprints in my way. This journey would not have been possible without the support of my professors, supervisors and mentors, family and friends. To my family thank you for encouraging me in all my pursuits and inspiring me to follow my dreams. -
Beginner's Guide to Asterisk
Getting Started With Open Source Telephony A Beginners Guide to Asterisk Steve Sokol Asterisk Marketing Lead, Digium Justin Hester Lead Asterisk Technical Instructor, Digium Agenda • Summary of Asterisk • Basics and Distributions • Resources • Asterisk is a toolkit • Distros are more complete • Training classes package with GUI • Digium and Asterisk • Getting started using Asterisk • AstriCon 2016 • Architecture - Linux + Asterisk • Difference between CLI and GUI • Versioning Getting Started with Asterisk Getting Started with Asterisk • Find it • Install it • Configure it What is Asterisk? An Open Source Communications Platform • Software, written in C, that you put on an ordinary operating system—transforming that system into a communications engine. Software – A communications platform A system through which communications flow, from one endpoint to another. What is Asterisk - Platform Open Source Communications Platform Software - Extensible Architecture A simple core with only a few responsibilities Module Management Reading Configuration CORE System Timing CORE Channel Management Software – Modular Architecture Modules app_dial.so • Use the native modules • Use Asterisk’s APIs to control and extend Asterisk – AMI, AGI, and ARI Preparing for Asterisk – Set up a host machine • Old physical hardware – Laptop, rackmount or tower system • Virtual machine – e.g. VirtualBox (on your Mac, Windows or Linux laptop!) www.virtualbox.org • Cloud server – e.g. www.digitalocean.com or aws.amazon.com Finding Asterisk – Choose your path ‘Source’ or ‘Plain Vanilla’ AsteriskNOW, a PBX ‘distro’. www.asterisk.org/downloads Choosing Asterisk - Source • Install a Linux operating system • Set up networking • Configure software repositories • Install Asterisk dependencies • Download and install Asterisk, DAHDI, LibPRI from provided scripts. And you will have an unconfigured, pristine, ready to configure “Asterisk Configuration Framework”. -
Integration of Asterisk IP-PBX with ESP32 Embedded System for Remote Code Execution †
Proceedings Integration of Asterisk IP-PBX with ESP32 Embedded System for Remote Code Execution † Juan Pablo Berrío López 1,* and Yury Montoya Pérez 2 1 Facultad de Ingeniería, Diseño e Innovación (FIDI), Institución Universitaria Politécnico Grancolombiano, 050034 Medellín, Colombia 2 Facultad de Ciencias Básicas e Ingeniería, Corporación Universitaria Remington, 050012 Medellín, Colombia * Correspondence: [email protected]; Tel.: +57-304-439-7353 † Presented at the 2nd XoveTIC Congress, A Coruña, Spain, 5–6 September 2019. Published: 5 August 2019 Abstract: 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. Keywords: sensors; wireless fidelity; internet of things; microcontrollers; ESP32; embedded systems 1. Introduction Internet of Things is a concept based on the connection of electronic devices with each other or through the Internet, which has been generating expectation for years as it is expected to be a great driver of digital transformation in homes and cities, as well as in companies. IP telephony has taken a central role in the information highway so that the network can interconnect each home and each business through a packet switching network [1,2]. In this work, the Asterisk pbx was used based on the Issabel Linux distribution, which allowed to create dial plans to receive calls from the users and through an IVR to make a POST request to the ESP32 platform, which within its code has established methods that for investigation, allowed to move a servo motor and turn on led lights. -
A Survey of Open Source Products for Building a SIP Communication Platform
Hindawi Publishing Corporation Advances in Multimedia Volume 2011, Article ID 372591, 21 pages doi:10.1155/2011/372591 Research Article A Survey of Open Source Products for Building a SIP Communication Platform Pavel Segec and Tatiana Kovacikova Department of InfoCom Networks, University of Zilina, Univerzitna 8215/1, 010 26 Zilina, Slovakia Correspondence should be addressed to Tatiana Kovacikova, [email protected] Received 29 July 2011; Revised 31 October 2011; Accepted 15 November 2011 Academic Editor: T. Turletti Copyright © 2011 P. Segec and T. Kovacikova. This is an open access article distributed under the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited. The Session Initiation Protocol (SIP) is a multimedia signalling protocol that has evolved into a widely adopted communication standard. The integration of SIP into existing IP networks has fostered IP networks becoming a convergence platform for both real- time and non-real-time multimedia communications. This converged platform integrates data, voice, video, presence, messaging, and conference services into a single network that offers new communication experiences for users. The open source community has contributed to SIP adoption through the development of open source software for both SIP clients and servers. In this paper, we provide a survey on open SIP systems that can be built using publically available software. We identify SIP features for service deve- lopment and programming, services and applications of a SIP-converged platform, and the most important technologies support- ing SIP functionalities. We propose an advanced converged IP communication platform that uses SIP for service delivery. -
Kolokované Nasazení Asterisk PBX a Kamailio SIP Proxy Ve Funkci SBC
VŠB – Technická univerzita Ostrava Fakulta elektrotechniky a informatiky Katedra telekomunikační techniky Kolokované nasazení Asterisk PBX a Kamailio SIP Proxy ve funkci SBC Combined Implementation of Asterisk PBX and SIP Proxy Kamailio in the Role of the SBC 2017 Bc. Róbert Šimovič Poďakovanie Rád by som poďakoval Ing. Janovi Rozhonovi, Ph.D., za odbornú pomoc a konzultáciu pri vytváraní tejto diplomovej práce. Abstrakt Práca sa zaoberá vytvorením SBC pomocou spoločného nasadenia pobočkovej ústredne Asterisk a SIP proxy Kamailio. Pri návrhu SBC bolo potrebné brať do úvahy spôsob implementácie oboch platforiem na spoločne zdieľanom virtuálnom serveri. Návrh je vystavaný na novších verziách platforiem a využíva tie výhody oboch platforiem, ktoré robia dané riešenie atraktívne. Kľúčové funkcie pre celý systém sú bezpečnosť a variabilita nastavenia SBC, ktoré je možné meniť podľa aktuálnych požiadaviek. V mojom realizovanom prostredí bolo potrebné navrhnúť súbor bezpečnostných funkcií, ktoré naplňovali funkciu SBC. Celý systém bol vystavený výkonnostnému testovaniu, ktorý som vyhodnocoval pomocou grafov. Zásadou metodiky testovania je možnosť reprodukovateľnosti. Kľúčové slová Kamailio, Asterisk, SBC, testovanie, SIPp, bezpečnosť, VoIP, HTABLE, PIKE, TOPOH, RATELIMIT, TLS Abstract This thesis deals with the creation of SBC by the combined implementation of Asterisk PBX and SIP Proxy Kamailio. When designing the SBC, it was necessary to take in consideration the implementation of both platforms on one share virtual server. The design is based on newer versions of mentioned platforms and uses the benefits of both platforms, which make the solution more attractive. The key functions of the whole system are security and variability of the SBC settings, which can be altered according to actual requirements. -
The 21St Century Business Telephone System
st System Application Accessories The 21 Century Business Telephone System snom D3 snom D7 Expansion Module Expansion Module IP Door Phone Video Conference System Polycom SoundStation snom C520 - WiMi snom C52 - SP IP Conference Phone IP Conference Phone snom C520 Extend Jabra Speak 510 USB Channel Bank 4/8 Ports FXO/FXS Cisco SPA112 expansion module VOIP Gateway 2 Ports FXS Gateway VOIP PBX Telephone System IP PBX CS-1200 (SIP) CS-2200 V2 CS-4200 Server Edition Model CS-1200 (SIP) CS-2200 V2 CS-4200 Server Edition snom DECT Solution Headsets Compact / 19” 1U Rack Mount 19” 2U Rack Mount 19” Rack Mount Chassis 19” 1U Rack Mount Recommended Capacity 25 Users 100 Users 300 Users 300 - 1000 Users Max. Concurrent Calls 15 50 200 500 snom M700 snom M65 snom M85 snom Headset Jabra Bluetooth Headset Max. PSTN Analog Ports - 32 128 256 - 2 8 8 Max. PSTN Digital Ports Please Contact : PSTN Port Type SIP CO / IDAP ( T1) / SIP CO / IDAP ( T1) / SIP CO / IDAP ( T1) / SIP Voice Recording Space 20 hours (Voice Mail) 13,000 hours 13,000 - 78,000 hours 13,000 hours or above Redundant Power Supply Unit - - - Available Contact Center - - Max 50 Agents Max 500 Agents Input Voltage 110 / 220V AC 110 / 220V AC 110 / 220V AC 110 / 220V AC Power Supply Unit Internal Internal Internal Internal www.platonvoip.com o o o o Operation Temperature 0 to 40 C 0 to 40 C 0 to 40 C 0 to 40 C © 2018 Style Online Ltd. All rights reserved PLATON® logo is registered trademark of Style Online Ltd. -
Fachhochschule Braunschweig/Wolfenbüttel University of Applied Sciences
View metadata, citation and similar papers at core.ac.uk brought to you by CORE provided by Universidad Carlos III de Madrid e-Archivo Fachhochschule Braunschweig/Wolfenbüttel University of Applied Sciences Proyecto Final de Carrera Editor de texto para el interfaz gráfico de AskoziaPBX Adriana Arroyo García Matrikel Nummer: 1008493 NIA: 100055633 Mentor : Prof. Diedrich Wermser Supervisor: Michael Iedema Julio 2009 Certifico que, excepto referencia expecífica, el trabajo descrito en este proyecto es original. Tampoco el proyecto en si, ni una parte de él, ha sido presentado con anterioridad en ninguna Universidad. La autora. Índice de contenido i Abstracto.............................................................................................................................................5 ii Agradecimientos................................................................................................................................6 1. Introducción......................................................................................................................................7 1.1 Motivación.................................................................................................................................7 1.2 Objetivos y tareas.......................................................................................................................8 2. Estado del arte.................................................................................................................................. 9 2.1 Introducción