Skype for Asterisk™ Administrator Manual

Total Page:16

File Type:pdf, Size:1020Kb

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 1.1 What is Asterisk®?..........................................................................................................9 1.2 Asterisk as a Phone Switch (PBX)..................................................................................9 1.3 Asterisk as a Gateway.....................................................................................................9 1.4 Asterisk as a Feature/Media Server................................................................................9 1.5 Asterisk in the Call Center.............................................................................................10 1.6 Asterisk in the Network..................................................................................................10 1.7 Asterisk Everywhere......................................................................................................10 Chapter 2: Installation............................................................................................................11 2.1 Installation Overview......................................................................................................12 2.2 Register Skype for Asterisk...........................................................................................13 2.2.1 Open Source Asterisk............................................................................................14 2.2.2 AsteriskNOW..........................................................................................................14 2.3 Install Skype for Asterisk...............................................................................................15 2.3.1 Open Source Asterisk............................................................................................15 2.3.2 AsteriskNOW..........................................................................................................16 2.4 Digium G.729 Software Codec Module.........................................................................17 2.5 Load Skype for Asterisk Modules..................................................................................18 2.6 Verify Installation...........................................................................................................20 2.7 Backup License File.......................................................................................................20 Chapter 3: Configuration........................................................................................................21 3.1 chan_skype.conf............................................................................................................22 3.2 Dial Technology.............................................................................................................26 3.2.1 Outgoing Calls........................................................................................................26 3.2.2 Incoming Calls........................................................................................................27 3.3 Functions.......................................................................................................................28 Digium, Inc. Page 3 3.3.1 Channel Function...................................................................................................28 3.3.2 CallerID Function....................................................................................................30 3.3.3 Skype Call Property Function.................................................................................31 3.3.4 Skype Account Property Function..........................................................................33 3.3.5 Skype Buddy Functions..........................................................................................36 3.3.6 Skype Chat Function..............................................................................................37 3.4 Hints...............................................................................................................................39 3.5 Manager Commands.....................................................................................................41 3.5.1 Skype Account Property.........................................................................................41 3.5.2 Skype Add Buddy...................................................................................................42 3.5.3 Skype Remove Buddy............................................................................................42 3.5.4 Skype Buddies.......................................................................................................43 3.5.5 Skype Buddy..........................................................................................................43 3.5.6 Skype Chat Send...................................................................................................44 Chapter 4: Troubleshooting...................................................................................................45 4.1 Manager Events.............................................................................................................45 4.1.1 Skype Account Status Events................................................................................45 4.1.2 Skype Buddy Status Events...................................................................................46 4.1.3 Skype Chat Message Events.................................................................................47 4.2 Asterisk Command Line Interface (CLI)........................................................................48 4.2.1 skype login user <username>................................................................................48 4.2.2 skype logoff user <username>...............................................................................48 4.2.3 skype show buddies <username>..........................................................................48 4.2.4 skype set debug off [user] [<username>]...............................................................49 4.2.5 skype set debug on [user] [<username>]...............................................................49 4.2.6 skype show hostid..................................................................................................49 4.2.7 skype show licenses...............................................................................................49 4.2.8 skype show settings...............................................................................................49 4.2.9 skype show users...................................................................................................49 4.2.10 skype show user <username>.............................................................................49 4.2.11 skype set vedebug off <username>.....................................................................49 Digium, Inc. Page 4 4.2.12 skype set vedebug on <username>.....................................................................49 4.2.13 skype show version..............................................................................................50 4.3 Frequently Asked Questions.........................................................................................51 Appendix A: Glossary and Acronyms..................................................................................58 Appendix B: DIGIUM END-USER PURCHASE AND LICENSE AGREEMENT....................69 Digium, Inc. Page 5 Chapter 1: Overview Digium's Skype for Asterisk™ (SfA) is an add-on channel driver for Asterisk based systems. Adding Skype for Asterisk to any Asterisk server enables complete access to the Skype community, including low cost PSTN access and free calling to over 440+ million Skype users. Skype for Asterisk integrates seamlessly with the Skype community. Skype for Asterisk performance is superior to the proxy or gateway products available for connecting to the Skype community. There
Recommended publications
  • 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
    [Show full text]
  • 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).
    [Show full text]
  • 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.
    [Show full text]
  • 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”.
    [Show full text]
  • 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.
    [Show full text]
  • 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.
    [Show full text]
  • 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
    [Show full text]
  • Put Skype on Every Phone in Your Office
    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.
    [Show full text]
  • 5 Culprits That Kill Asterisk® IP-PBX Performance
    5 Culprits that Kill Asterisk® IP-PBX Performance …and what you can do about them www.xorcom.com Asterisk is a registered trademark of Digium, Inc. Agenda . Goal of Webinar . Xorcom Credentials . The 5 Culprits . Load Test Results . Guidelines for Best Performance . Questions & Answers www.xorcom.com Common Misconception . Choosing a high-priced server will give you the best performance You will pay too much for an unsuitable solution www.xorcom.com Goal of the Webinar . Clarification . This discussion relates to workhorse PBX, not residential or demo systems . Review parameters that affect performance . Criteria for choosing hardware . Ways to optimize software . Pass along lessons learned www.xorcom.com Xorcom Credentials . Established in 2004 . All products are based on Asterisk . Xorcom drivers: standard component in Asterisk since v. 1.2.4 (Feb ‘06) . Award-winning, flexible and modular telephony interface solutions based on XPP™ technology (USB2) . Emphasis on built-in reliability www.xorcom.com About IP Gateways… Feature Astribank IP Gateway Central management via Asterisk IP-PBX LAN-independent architecture Fully Asterisk integrated Redundant power supply Reliable fax support Auto detection by Asterisk IP-PBX Auto configuration by Asterisk IP-PBX Low Cost of Ownership (COO) www.xorcom.com So…what are the 5 culprits? 1. CPU 2. Firmware-Motherboard Design 3. Chipset 4. Peripherals 5. Application www.xorcom.com 1. CPU . Speed . Disable CPU Hyper-Threading when you have many Astribanks connected . Number of Cores . Additional cores are beneficial, especially when additional applications (such as call centers) are run on top of Asterisk . MMU (Memory Management Unit) performance . Some processors under test have demonstrated bad MMU performance www.xorcom.com 2.
    [Show full text]
  • VOICE OVER IP with ASTERISK 11 Postgresql-Based CDR and Saw Multiple Entries of the Same Toll-Free Number
    ASTERISK IS A SOFTWARE-BASED PBX that runs under Linux, *BSDs, and Mac OS X. HEISON CHAK It can also be used as a VoIP gateway and can provide a bridge to the PSTN (Public Switched Telephone Network, a.k.a. Plain Old Telephone Service). IAX (Inter-Asterisk voice over IP eXchange), pronounced “eeks,” is a protocol with Asterisk used by Asterisk as an alternative to SIP, H.323, etc., when connecting to other VoIP Heison Chak is a system and network administrator devices that support IAX. at SOMA Networks. He focuses on network manage- ment and performance analysis of data and voice networks. Heison has been an active member of the Asterisk community since 2003 and will be deliver- This article describes some of the interesting things I ing a tutorial on VoIP Principles and Practice at the have done using Asterisk and VoIP, told as the story of USENIX ’05 Annual Technical Conference. a business trip from my home in Toronto. [email protected] It was 12:15 in the morning and I had just gotten home from work. Realizing that I had not yet packed for my trip to San Francisco, I was a little worried about missing my early morning flight. I knew that snooze alarms no longer worked for me; I would often keep hitting the snooze button when the alarm went off, or I would just be so used to the sound that I wouldn’t even hear it at all. Either way, I could wind up being late. The AGI (Asterisk Gateway Interface) script I had installed on my Asterisk box that serves wake-up calls solved both my problems with snooze alarms.
    [Show full text]
  • Introduction to the Asterisk Open Source PBX
    Introduction to the Asterisk Open Source PBX http://asterisk.org Mark Spencer Linux Support Services, Inc. http://www.linux-support.net Presented first at Libre Software Meeting 2002 Bordeaux, France July 9, 2002 Page 1/16 Introduction to Asterisk Asterisk is a fully Open Source, hybrid TDM and packet voice PBX and IVR platform. Asterisk is and has been Open Source under GNU GPL (with an exception permitted for linking with the OpenH323 project, in order to provide H.323 support). Commercial licensing is available from Linux Support Services, Inc. (http://www.linux-support.net) for applications in which the GPL is inappropriate. Unlike many modern "soft switches", Asterisk can use both traditional TDM technology and packet voice (Voice over IP and Voice over Frame Relay) protocols. Calls switched on TDM interfaces provide lag-less TDM call quality, while retaining interoperability with VoIP packetized protocols. Asterisk acts as a full featured PBX, supporting virtually all conventional call features on station interfaces, such as Caller*ID, Call Waiting,Caller*ID on Call Waiting, Call Forward/Busy, Call Forward/No Answer, Call Forward Variable, Stutter Dialtone, Three-way Calling, Supervised Transfer, Unsupervised Transfer, ADSI enhancements, Voicemail, Meet-me Conferencing, Least Cost Routing, VoIP gatewaying, Call Detail Records, etc. At the same time, Asterisk provides full IVR capability, programmable at several layers, from a low-level C interface, to high level AGI scripting (analogous to CGI) and extension logic interfaces. Asterisk IVR applications run seamlessly from one interface to another, and need not know anything about the physical interface, protocol, or codec of the call they are working with, since Asterisk provides total abstraction for all those concepts.
    [Show full text]
  • Name Description
    SoX(7) Sound eXchange SoX(7) NAME SoX − Sound eXchange, the Swiss Army knife of audio manipulation DESCRIPTION This manual describes SoX supported file formats and audio device types; the SoX manual set starts with sox(1). Format types that can SoX can determine by a filename extension are listed with their names preceded by a dot. Format types that are optionally built into SoX are marked ‘(optional)’. Format types that can be handled by an external library via an optional pseudo file type (currently sndfile) are marked e.g. ‘(also with −t sndfile)’. This might be useful if you have a file that doesn’twork with SoX’sdefault format readers and writers, and there’sanexternal reader or writer for that format. To see if SoX has support for an optional format or device, enter sox −h and look for its name under the list: ‘AUDIO FILE FORMATS’ or ‘AUDIO DEVICE DRIVERS’. SOXFORMATS & DEVICE DRIVERS .raw (also with −t sndfile), .f32, .f64, .s8, .s16, .s24, .s32, .u8, .u16, .u24, .u32, .ul, .al, .lu, .la Raw(headerless) audio files. For raw,the sample rate and the data encoding must be givenusing command-line format options; for the other listed types, the sample rate defaults to 8kHz (but may be overridden), and the data encoding is defined by the givensuffix. Thus f32 and f64 indicate files encoded as 32 and 64-bit (IEEE single and double precision) floating point PCM respectively; s8, s16, s24,and s32 indicate 8, 16, 24, and 32-bit signed integer PCM respectively; u8, u16, u24, and u32 indicate 8, 16, 24, and 32-bit unsigned integer PCM respectively; ul indicates ‘µ-law’ (8-bit), al indicates ‘A-law’ (8-bit), and lu and la are inverse bit order ‘µ-law’ and inverse bit order ‘A-law’ respectively.For all rawformats, the number of channels defaults to 1 (but may be over- ridden).
    [Show full text]