DOCSLIB.ORG

Put Skype on Every Phone in Your Office

Total Page：16

File Type：pdf, Size：1020Kb

Download full-text PDF Read full-text

Load more

PUT SKYPE ON EVERY PHONE IN YOUR OFFICE. Skype certified User friendly This ensures you get the highest Easy to set up and use without quality experience when making requiring IT expertise. Skype calls. Full featured Extremely cost effective Compares favorably to high-end A fraction of the cost of enterprise-class systems. comparable systems. Imagine the money you’d save if you were able to seamlessly make calls using Skype from any phone in your office. Imagine being able to do this with a phone system that is a fraction of the cost of comparable systems. Meet FREETALK Connect. Setup & Management Auto-Provisioning of IP Wizard Driven Setup Remote Management Phones The FREETALK Connect is designed If your FREETALK Connect system Just by connecting your IP desktop for the do-it-yourselfer. The wizard administrator or support resource is and conferencing phones to your walks users through a series of located somewhere that does not network, your phones will be simple questions about their permit direct access to your detected and configuration files for organization and uses this FREETALK Connect appliance, that’s your phones generated and pushed information to automatically no problem. FREETALK Connect has out to your phones. FREETALK configure the system for basic remote administration capabilities Connect auto-provisioning works communications management within that enable the system to be with almost all models of desktop the organization. The wizard installs administered from anywhere Internet and conference IP phones from and configures all basic networking, access is available. providers such as Aastra, Cisco telephony system and user Small Business Pro series, Linksys, functionality on the FREETALK Polycom and snom. Connect and requires no IT or telephony expertise or experience. Report Exporting (.csv) Status Pages Extend FREETALK Connect’s powerful reporting View the real-time status of the FREETALK Connect engine, including Call Detail Reports, by creating system. You can quickly see which phones are .csv files that you can import into applications connected and which analog and VoIP phone such as Excel and Access. services are up and running. Networking WAN Connectivity LAN Services Routing FREETALK Connect provides Wide Local Area Network (LAN) services FREETALK Connect incorporates a Area Network (WAN) connectivity for all such as DHCP server, static DHCP, complete network router supporting standard broadband connection DNS caching and static DNS naming NAT addressing, port forwarding, services. FREETALK Connect supports are supported and easily managed by Universal Plug and Play (UPnP) and always-on connections using DHCP FREETALK Connect. quality of service (QoS). (typical of cable and some DSL connections); username and password supported systems (PPPoE, typically Dynamic DNS used by DSL connections); or Internet Dynamic DNS is supported for most connections requiring a static IP popular dynamic DNS services address. (DynDNS, etc.) Skype Skype Enabled Receive Inbound Make Outbound Click-to-Call Skype Manager FREETALK Connect Calls Calls Use the FREETALK Use Skype Manager to enables you to receive Accept calls via a Make outbound calls Connect SkypeID to establish and manage and make calls via Skype for Business ID using SkypeOut™ with create click-to-call Skype for Business Skype on every phone or a Skype Online™ low rates worldwide. services on Websites accounts for users, to in your office. Easily number. or email signatures. purchase Skype integrate with Credits and allocate to supported IP phones. users, and to view call detail reports for Skype calling activity. Telephony #3 #2 #1 Auto-Attendant (IVR) Voicemail The FREETALK Connect Auto-Attendant lets callers press FREETALK Connect offers easy ways to retrieve your voicemail options to reach destination extensions within your company. messages: For example, "Press 1 for Support" or "Press 2 for Sales." The Auto-Attendant feature is easy to use and powerful. With a few simple clicks of the mouse you can manage call flow, forward calls off-site, and more. options. 1 3 2 4 Instant Number Portability Attach up to four existing PSTN lines and let the FREETALK Connect manage your existing phone numbers. Easily Unified Voicemail-to-Email transition into the additional features and benefits offered by Receive your voicemails as emails! FREETALK Connect can FREETALK Connect and its Skype enhancements. send an email to the voicemail recipient whenever they receive a voicemail. The email has an audio attachment so that you can listen to the voicemail right from your Inbox. Alternatively, FREETALK Connect synchronizes messages with your IMAP server to unify your inbox. If the message is opened or deleted in email it will also be opened or deleted from the phone and vice versa. Telephony Unlimited VoIP Accounts IP Phones Distinctive Ring In addition to Skype, FREETALK FREETALK Connect is one of the most Know who is being called or what type Connect comes VoIP-ready with no flexible phone systems on the market of call is coming in by the distinctive limit to the number of third party VoIP today, supporting all analog phones ring. In a small business the same service provider accounts you can and auto-provisioning popular IP person may perform many roles. setup. In a few minutes, you can phones such as Aastra, Cisco Small Distinctive rings will identify if the call is configureFREETALK Connect to Business, Linksys, Polycom, and coming in as a result of someone include your favorite VoIP Providers snom. FREETALK Connect supports selecting the sales option from the (SIP) and configure VoIP for inter-office MWI (Message Waiting Indicator) on all auto-attendant or if it's a support call. In and intra-office calls while keeping IP Phones. the home, distinctive rings can identify regular calls going out over PSTN or who the inbound call is for. Skype. Note: Not all phones support customizable ring tones. Simultaneous Ring Call Forwarding Take one inbound call and ring all phones or those identified Users can use their personal User Portal to enable call as part of a Team; the first to pick up gets the call. forwarding to either an internal extension or to an external number. #3 #2 #1 Name Directory Music/Messages on Hold FREETALK Connect comes pre-configured with a Music-on-Hold (MOH) is as simple as uploading audio files professional "spell-by-name" directory. Callers are simply using the Web-based administration panel. The audio files prompted to "spell the first three letters of the party's first may contain music or messages. You choose what you would name." They are then automatically connected to the like to have your callers listen to while on hold. requested extension. Least Cost Routing PSTN Failover FREETALK Connect can configure outbound routes such that FREETALK Connect offers VoIP-users a PSTN failover in case local calls are prioritized to go out over local fixed-rate PSTN of an Internet service interruption. PSTN Failover ensures that analog lines while long distance calls are routed via SkypeOut if your Internet service goes down, your PSTN phone line is or an alternative VoIP service provider at low per-minute still available and operational. calling rates available via Skype or the VoIP service provider. Telephony Remote User / Office User Portal Find Me FREETALK Connect enables remote FREETALK Connect User Portal Allow callers to find you wherever you are. users/offices to work as though they enables users to manage their phone Simply indicate how you would like to be were in the same location. Calls can system functions from an easy to use reached whether at home, a branch office, come in through the main phone number interface from their PC rather than using another internal extension or on your mobile and will be routed to them no matter the phone keypad to manage their phone. Find Me capabilities will allow where they are located. Extension to personal phone preferences. The User FREETALK Connect to find you by ringing extension dialing works as if the remote Portal allows users to quickly and easily other numbers or extensions sequentially or user is just down the hall. Using manage their personal activities such can even ring multiple phones soft-phones or Wi-Fi enabled mobile as: listen to voicemail, enable simultaneously. phones with a SIP client ensures the road do-not-disturb (DND), enable call warrior and frequent traveler is always in forwarding, change their email address touch. IP phones, once provisioned, can for voicemail to email, user PIN and optionally connect via any Internet personal password from anywhere they connection worldwide. have Internet access. Teams Paging Custom Caller IDs FREETALK Connect includes a powerful Want to page everyone on your phone FREETALK Connect lets you customize Team feature that allows you to build system? Simply dial *00 on your phone the Caller ID name/number for each multi-user Team extensions and assign (or pre-programmed speed dial key) user and team allowing you to block, permissions to those Teams. You can and broadcast your message or request reveal, or change the Caller ID of every control Team permissions such as long to all phones in the system. Note: There user or team on your system. Note: if distance dialing (outbound line control), are a few phones that cannot receive you are using analog PSTN lines, for distinctive ring tones, call forwarding pages. Check your phone's user 911 reasons, PSTN outbound Caller IDs and unique callerID. manual. are always controlled by your carrier. Routing by DID’s or CallerID Advanced Call Forwarding Conference Bridge With FREETALK Connect’s ability to With a few clicks of your mouse in the FREETALK Connect comes route by DID’s or CallerID’s, you can User Portal you can forward your pre-configured with one conference direct calls to a different call menu extension to another extension, a bridge ready to use for free! Up to ten based on the inbound number dialed or mobile phone, or any other number.
Recommended publications
  • Skype for Asterisk™ Administrator Manual

    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
  • Internet Telephony Over Wireless Links

    Internet Telephony Over Wireless Links

    Internet Telephony over Wireless Links vorgelegt von Diplom-Ingenieur Christian Hoene von der Fakult¨at IV - Elektrotechnik und Informatik der Technischen Universit¨at Berlin zur Erlangung des akademischen Grades Doktor der Ingenieurwissenschaften – Dr.-Ing. – genehmigte Dissertation Promotionsausschuss: Vorsitzender: Prof. Dr. Heiß Berichter: Prof. Dr.-Ing. Wolisz Berichter: Prof. Dr.-Ing. Steinmetz Berichter: Prof. Dr.-Ing. Sikora Tag der wissenschaftlichen Aussprache: 16. Dezember 2005 Berlin 2006 D83 c 2005-2006 by Christian Hoene Jahnstr. 22 72144 Dußlingen Germany [email protected] As a small child, you never spoke that clearly, no wonder that you want to improve the speech perceptibility. My mother, after hearing my thesis topic. Abstract This thesis presents algorithms to enhance the efficiency of packetized, interactive speech communication over wireless networks. The results achieved are the following: We present an improved approach to assess the quality of voice transmissions in IP-based communication networks. We combined the ITU E-Model, the ITU PESQ algorithm, and various codec and playout schedulers to analyse VoIP traces. Parts of this algorithm have been included in ITU standards. By using this assessment approach we derived design guidelines for application and data-link protocols. Also, we developed a quality model to parametrise adaptive VoIP applications. Later results received a best-paper award. If highly compressed packetized speech is transported over packet networks, losses of in- dividual packets impair the perceptual quality of the received stream differently, depending on the content and context of the lost packets. We introduce the idea of the Importance of Individual Packets, which is defined by the impact of VoIP packet loss on speech quality.
  • Sangoma T116 16-Span T1/E1/J1 Tapping Board

    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
  • ATA User's Manual

    ATA User's Manual

    VoIP Analog Telephone Adapter VIP-156 VIP-157 User’s manual 1 Copyright Copyright (C) 2006 PLANET Technology Corp. All rights reserved. The products and programs described in this User’s Manual are licensed products of PLANET Technology, This User’s Manual contains proprietary information protected by copyright, and this User’s Manual and all accompanying hardware, software, and documentation are copyrighted. No part of this User’s Manual may be copied, photocopied, reproduced, translated, or reduced to any electronic medium or machine-readable form by any means by electronic or mechanical. Including photocopying, recording, or information storage and retrieval systems, for any purpose other than the purchaser's personal use, and without the prior express written permission of PLANET Technology. Disclaimer PLANET Technology does not warrant that the hardware will work properly in all environments and applications, and makes no warranty and representation, either implied or expressed, with respect to the quality, performance, merchantability, or fitness for a particular purpose. PLANET has made every effort to ensure that this User’s Manual is accurate; PLANET disclaims liability for any inaccuracies or omissions that may have occurred. Information in this User’s Manual is subject to change without notice and does not represent a commitment on the part of PLANET. PLANET assumes no responsibility for any inaccuracies that may be contained in this User’s Manual. PLANET makes no commitment to update or keep current the information in this User’s Manual, and reserves the right to make improvements to this User’s Manual and/or to the products described in this User’s Manual, at any time without notice.
  • Voip Tutorial

    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).
  • VOICE OVER INTERNET PROTOCOL (Voip)

    VOICE OVER INTERNET PROTOCOL (Voip)

    S. HRG. 108–1027 VOICE OVER INTERNET PROTOCOL (VoIP) HEARING BEFORE THE COMMITTEE ON COMMERCE, SCIENCE, AND TRANSPORTATION UNITED STATES SENATE ONE HUNDRED EIGHTH CONGRESS SECOND SESSION FEBRUARY 24, 2004 Printed for the use of the Committee on Commerce, Science, and Transportation ( U.S. GOVERNMENT PUBLISHING OFFICE 22–462 PDF WASHINGTON : 2016 For sale by the Superintendent of Documents, U.S. Government Publishing Office Internet: bookstore.gpo.gov Phone: toll free (866) 512–1800; DC area (202) 512–1800 Fax: (202) 512–2104 Mail: Stop IDCC, Washington, DC 20402–0001 VerDate Nov 24 2008 14:00 Dec 07, 2016 Jkt 075679 PO 00000 Frm 00001 Fmt 5011 Sfmt 5011 S:\GPO\DOCS\22462.TXT JACKIE SENATE COMMITTEE ON COMMERCE, SCIENCE, AND TRANSPORTATION ONE HUNDRED EIGHTH CONGRESS SECOND SESSION JOHN MCCAIN, Arizona, Chairman TED STEVENS, Alaska ERNEST F. HOLLINGS, South Carolina, CONRAD BURNS, Montana Ranking TRENT LOTT, Mississippi DANIEL K. INOUYE, Hawaii KAY BAILEY HUTCHISON, Texas JOHN D. ROCKEFELLER IV, West Virginia OLYMPIA J. SNOWE, Maine JOHN F. KERRY, Massachusetts SAM BROWNBACK, Kansas JOHN B. BREAUX, Louisiana GORDON H. SMITH, Oregon BYRON L. DORGAN, North Dakota PETER G. FITZGERALD, Illinois RON WYDEN, Oregon JOHN ENSIGN, Nevada BARBARA BOXER, California GEORGE ALLEN, Virginia BILL NELSON, Florida JOHN E. SUNUNU, New Hampshire MARIA CANTWELL, Washington FRANK R. LAUTENBERG, New Jersey JEANNE BUMPUS, Republican Staff Director and General Counsel ROBERT W. CHAMBERLIN, Republican Chief Counsel KEVIN D. KAYES, Democratic Staff Director and Chief Counsel GREGG ELIAS, Democratic General Counsel (II) VerDate Nov 24 2008 14:00 Dec 07, 2016 Jkt 075679 PO 00000 Frm 00002 Fmt 5904 Sfmt 5904 S:\GPO\DOCS\22462.TXT JACKIE C O N T E N T S Page Hearing held on February 24, 2004 ......................................................................
  • Authenticall: Efficient Identity and Content Authentication for Phone

    Authenticall: Efficient Identity and Content Authentication for Phone

    AuthentiCall: Efficient Identity and Content Authentication for Phone Calls Bradley Reaves, North Carolina State University; Logan Blue, Hadi Abdullah, Luis Vargas, Patrick Traynor, and Thomas Shrimpton, University of Florida https://www.usenix.org/conference/usenixsecurity17/technical-sessions/presentation/reaves This paper is included in the Proceedings of the 26th USENIX Security Symposium August 16–18, 2017 • Vancouver, BC, Canada ISBN 978-1-931971-40-9 Open access to the Proceedings of the 26th USENIX Security Symposium is sponsored by USENIX AuthentiCall: Efficient Identity and Content Authentication for Phone Calls Bradley Reaves Logan Blue Hadi Abdullah North Carolina State University University of Florida University of Florida reaves@ufl.edu bluel@ufl.edu hadi10102@ufl.edu Luis Vargas Patrick Traynor Thomas Shrimpton University of Florida University of Florida University of Florida lfvargas14@ufl.edu [email protected]fl.edu [email protected]fl.edu Abstract interact call account owners. Power grid operators who detect phase synchronization problems requiring Phones are used to confirm some of our most sensi- careful remediation speak on the phone with engineers tive transactions. From coordination between energy in adjacent networks. Even the Federal Emergency providers in the power grid to corroboration of high- Management Agency (FEMA) recommends that citizens value transfers with a financial institution, we rely on in disaster areas rely on phones to communicate sensitive telephony to serve as a trustworthy communications identity information (e.g., social security numbers) to path. However, such trust is not well placed given the assist in recovery [29]. In all of these cases, participants widespread understanding of telephony’s inability to depend on telephony networks to help them validate provide end-to-end authentication between callers.
  • THE BENEFITS of VOIP for SMALL- to MEDIUM-SIZED BUSINESSES 3 Other Considerations There Are Many Costs to Consider for Expanding

    THE BENEFITS of VOIP for SMALL- to MEDIUM-SIZED BUSINESSES 3 Other Considerations There Are Many Costs to Consider for Expanding

    The Benefits of VoIP for Small- to Medium-Sized Businesses How utilizing the latest VoIP technologies can reduce maintenance costs while contributing to improved retention and growth within a business Over the past several years, we have seen the impact of mobile technology in our personal lives and how we shop for products and services. While businesses continue to make improvements to their websites, customer service Gone are the days of management platforms, and accounting and operational depending on a single software, their phone systems are overlooked for operational strand of copper wire efficiencies. This white paper examines why businesses need for your entire business to consider upgrading their communication systems to VoIP, communications. With and how utilizing the latest VoIP technologies can reduce hosted voice your maintenance costs while contributing to improved retention business will never miss and growth within a business. an opportunity or any client communication. Existing Phone Systems Hosted Voice is scalable to your company’s Businesses typically invested in a traditional phone system through the purchase of growing needs, without phone lines from their local telecom representative, and then purchasing a physical phone management system that was mounted in a closet near the company servers. To make downtime or anyone extension changes, a tech from the phone company or an IT network engineer would noticing that changes physically re-route the phone wire from one point on the “punch” board to another. This took have been made. It’s time and scheduling to move the phone and have a person available to perform the action.
  • Asterisk Record Codec

    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.
  • Beginner's Guide to Asterisk

    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”.
  • SIP Voip Phone VOI-7000

    SIP Voip Phone VOI-7000

    SIP VoIP Phone VOI-7000 H/W Version: 1 The VOI-7000 has all the features of a normal Simplicity in one device VoIP phone but with more functionality where The VOI-7000 compatibility with all SIP and VoIP different media can now be transferred in one service providers. Also allows for 3 different session at the same time. SIP accounts (IP telephone numbers) to be Digital Telephony in one device used from a single phone. The QoS (Quality of Service) function to ensures that clear The VOI-7000 from LevelOne is a feature-rich communication is possible at all times regardless yet cost competitive with high quality voice, of the LAN load. speakerphone and a 2x16 LCD screen that displays caller ID, number dialed, speed dial From one phone to a phone network entries. The keypad also doubles as an input The VOI-7000 is a standalone VoIP SIP phone, device to set up your SIP account and to connect but combined with other phones in a network to the Internet. by using an IP PBX at different branch offices and gateways to connect the IP network The numeric pad is tone activated so you can with traditional analogue phone systems, use it with telephone services that incorporates enables significant cost savings when making interactive telephony systems. Also support international or long distance phone calls. If volume adjustment, 10 button speed dial, using LevelOne IP PBX systems, it is possible phone book setup and programming, redial, call to assign extension numbers to a large phone hold, call wait and forwarding.
  • Integration of Asterisk IP-PBX with ESP32 Embedded System for Remote Code Execution †

    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.