I had the fantastic opportunity to work as the community manager for the Asterisk project at Digium for two years, which gave me one of the best vantage points for seeing the scope and imagination of the worldwide development effort pushing Asterisk forward. The depth and breadth of Asterisk is staggering—installations with hundreds of thousands of users are now commonplace.

I see Asterisk making deep inroads into the financial, military, hospital, Fortune enterprise, service provider, calling card, and mobile environments.

Asterisk has been emblematic of the way that open source software has changed business—and changed the world. My favorite part of any Asterisk project overview or conference talk is answering questions from someone new to Asterisk. Streaming MP3s into or out of phone calls?

Emailing recorded conference calls to the participants? Integration of voice services into existing Java apps? Yes, yes, yes, yes. The affirmative answers just keep flowing, and at that point, the best thing to do is to sit the person down and start showing him quick demonstrations of how Asterisk can be quickly deployed and developed. Then, I typically point the person toward the first edition of this book, Asterisk: The Future of Telephonyand set him loose.

In just a few hours of development or longer, of coursecompanies can change the way they deliver products to customers, nonprofits can overhaul how their users interact with the services they offer, and individuals can learn to build a perfectly customized call-handling system for their mobile and home phones.

Asterisk scales up and down from individual lines to vast multiserver installations across multiple continents, but the way to start is to install the package, open up some of the configuration files, and start looking at examples. From the basic beginnings of a PBX that Mark Spencer coded inthe Asterisk project, with the help of thousands of developers, has moved from simply connecting phone calls and has matured into a platform that can handle voice, video, and text across dozens of virtual and physical interface types.

The creation and growth of Asterisk were the inescapable results of the convergence of the four horsemen of the proprietary hardware apocalypse: Even hardware vendors who may be frightened of Asterisk from a competitive standpoint are using it in their labs and core networks: Nearly 75 percent raised their hands.

Asterisk is a mature, robust software platform that permeates nearly every area of the telecommunications industry and has firmly cemented itself as one of the basic elements in any open source service delivery system. Asterisk is accessible because of the ease with which a novice can understand basic concepts. But the first step for anyone, no matter what his or her skill level, is to look at examples of basic apps others have written.

For those of you building the most complex installations, there is even more interesting work—which will be realized quite soon—in development. The currently-in-development Asterisk SCF Scalable Communications Framework is being built as an adjunct open source project to allow Asterisk 1. Open source projects are hungry for new ideas and excellent contributions: I encourage you to be a participant in the Asterisk community, and I look forward to seeing your questions and examples in the next edition of this book.

Asterisk is an open source, converged telephony platform, which is designed primarily to run on Linux. Asterisk combines more than years of telephony knowledge into a robust suite of tightly integrated telecommunications applications. The power of Asterisk lies in its customizable nature, complemented by unmatched standards compliance. No other PBX can be deployed in so many creative ways. Applications such as voicemail, hosted conferencing, call queuing and agents, music on hold, and call parking are all standard features built right into the software.

Moreover, Asterisk can integrate with other business technologies in ways that closed, proprietary PBXs can scarcely dream of.

Asterisk can appear quite daunting and complex to a new user, which is why documentation is so important to its growth. Documentation lowers the barrier to entry and helps people contemplate the possibilities.

The Definitive Guide is the third edition of what was formerly called Asterisk: The Future of Telephony. We decided to change the name because Asterisk has been so wildly successful that it is no longer an up-and-coming technology. This book was written for, and by, members of the Asterisk community. However, this book will also be useful for the more experienced Asterisk administrator.

This is where we chop up the kindling and light the fire. Discusses the file structure of an Asterisk system. Covers obtaining, compiling, and installing Asterisk. Describes some initial configuration tasks for your new Asterisk system. This chapter goes over some of the configuration files required for all Asterisk installations. Provides guidance on configuring Asterisk to allow devices such as telephones to connect and make calls.

Introduces the heart of Asterisk, the dialplan. Discusses how to configure Asterisk to connect to other systems, such as other Asterisk servers, Internet telephony service providers, or the plain old telephone network. Covers the usage of one of the most popular applications included with Asterisk, the voicemail system.

Focuses on issues that an Asterisk administrator should be aware of when deploying a system outside of North America. Goes over some more advanced dialplan concepts. Describes the usage of two popular telephony features included with Asterisk, call parking and paging.

Covers techniques for routing calls between different administrative domains on the Internet. Discusses how to build call queues in Asterisk. Introduces the concept of device states and how they can be used as presence indicators. Covers how to build a menuing system using the Asterisk dialplan. Discusses various ways that Asterisk can be integrated with a database.

Goes over how Asterisk can be used to build applications that act on input provided by a caller. Provides instructions on how to connect to external services including LDAP, calendars, IMAP for voicemail, XMPP, Skype, and text-to-speech. Discusses the various options for integrating sending and receiving faxes with an Asterisk system. Introduces a network API for monitoring and controlling an Asterisk system. Introduces the Asterisk API that allows call control to be implemented in any programming language.

Discusses a number of approaches for clustering multiple Asterisk servers together once the demands of a deployment exceed the capabilities of a single server. Covers a peer-to-peer protocol native to Asterisk that can be used for call routing. Introduces some of the interfaces available for logging and monitoring an Asterisk system. A survey of some of the web interfaces that complement an Asterisk installation.

Discusses some common security issues that Asterisk administrators should be aware of. In conclusion, we discuss some of the things we expect to see from open source telephony in the near future. Explores the technologies in use in traditional telecom networks.

This used to be a chapter in old versions of this book. Delves into all the particularities of Voice over IP. This was also a chapter in old versions of this book. Contains information you should be aware of and take into consideration when planning an Asterisk deployment.

This book is focused on documenting Asterisk version 1. Linux is the operating system we have run and tested Asterisk on, and we have documented installation instructions for both CentOS Red Hat Enterprise Linux—based and Ubuntu Debian-based where they differ from each other. Indicates new terms, URLs, email addresses, filenames, file extensions, pathnames, directories, and package names, as well as Unix utilities, commands, options, parametersand arguments. Used to display code samples, file contents, command-line interactions, library names, and database commands.

Indicates commands or other text that should be typed literally by the user. Also used for emphasis in code. Shows text that should be replaced with user-supplied values. Indicates optional keywords and arguments. Signifies either choice-1 or choice This book is here to help you get your job done. In general, you may use the code in this book in your programs and documentation. For example, writing a program that uses several chunks of code from this book does not require permission.

Answering a question by citing this book and quoting example code does not require permission. We appreciate, but do not require, attribution. An attribution usually includes the title, author, publisher, and ISBN. Copyright Leif Madsen, Jim Van Meggelen, and Russell Bryant, Try it for free at http: Please address comments and questions concerning this book to the publisher:. We have a web page for this book, where we list errata, examples, and any additional information.

You can access this page at:. To comment or ask technical questions about this book, send email to:. Find us on Facebook: Follow us on Twitter: Watch us on YouTube: To David Duffett, thanks for the excellent chapter on internationalization, which would not have been served well by being written by us North Americans. Next, we want to thank our fantastic editor, Michael Loukides, for your patience with this third edition, which took too long to get off the ground, and many long months to finally get written.

Mike offered invaluable feedback and found incredibly tactful ways to tell us to rewrite a section or chapter when it was needed, and make us think it was our idea.

Mike built us up when we were down, and brought us back to earth when we got uppity. Thanks also to Rachel Head nee Rachel Wheelerour copyeditor, who fixes all our silly grammar, spelling, and style mistakes and the many Canadianisms that Leif and Jim feel compelled to includeand somehow leaves the result reading as if it was what we wrote in the first place.

Copyeditors are the unsung heroes of publishing, and Rachel is one of the very best. During the course of writing this book, we had the pleasure of being able to consult with many people with specific experience in various areas. Their generous contributions of time and expertise were instrumental in our research. Thanks to Randy Resnick, organizer of the VoIP User Group; Kevin Fleming of Digium; Lee Howard, author of iaxmodem and hylafax; Joshua Colp of Digium; Phillip Mullis of the Toronto Asterisk Users Group; Allison Smith, the Voice of Asterisk; Flavio E.

Goncalves, author of books on Asterisk, OpenSER, and OpenSIPS; J. Oquendo, Security Guru; Tzafrir Cohen, font of knowledge about security and lots of other stuff; Jeff Gehlbach, for SNMP; Ovidiu Sas, for your encyclopedic knowlege of SIP; Tomo Takebe, for some SMDI help; Steve Underwood, for help with fax and spandsp ; and Richard Genthner and John Covert, for helping with LDAP. A special thanks should also go to John Todd for being one of the first to write comprehensive Asterisk how-tos, all those years ago, and for all the many other things you do and have done for the Asterisk community.

Community members were able to submit feedback and comments, which was of enormous help to us. The following is a list of their names or handles [ 1 ]:. Matthew McAughan, Matt Pusateri, David Van Ginneken, Asterisk Mania, Giovanni Vallesi, Mark Petersen, thp4, David Row, tvc, Frederic Jean, John Todd, Steven Sokol, Laurent Steffan, Robert Dailey, Howard Harper, Joseph Rensin, Howard White, Jay Eames, Vincent Thomasset, Dave Barnow, Sebastien Dionne, Igor Nikolaev, Arend van der Kolk, Anwar Hossain, craigesmith, nkabir, anest, Nicholas Barnes, Alex Neuman, Justin Korkiner, Stefan Schmidt, pabelanger, jfinstrom, roderickmontgomery, Shae Erisson, Gaston Draque, Richard GenthnerMichael S Collins, and Jeff Peeler.

Thanks to Sean Bright, Ed Guy, Simon Ditner, and Paul Belanger for assisting us with clarifying best practices for user and group policies for Asterisk installation. In the past it was common to just install Asterisk with root permissions, but we have elected to describe an installation process that is more in keeping with Linux best practices, [ 2 ] and these fine gents contributed to our discussions on that.

Kudos to all the folks working on the FreeSWITCH, YATE, SER, Kamailio, OpenSIPS, SER, sipXecs, Woomera, and any other open source telecom projects, for stimulating new thoughts, and for pushing the envelope. Everyone in the Asterisk community also needs to thank Jim Dixon for creating the first open source telephony hardware interfaces, starting the revolution, and giving his creations to the community at large. Finally, and most importantly, thanks go to Mark Spencer, the original author of Asterisk and founder of Digium, for Asterisk, for Pidgin http: Asterisk is your legacy!

It worked quite well, and allowed us to play video games while conversing with each other—at least, until we wanted to add a third participant.

I turned to IRC and explained what I was looking for. Someone I wish I knew who mentioned that I should check out some software called Asterisk he presumably must have thought I was looking for MeetMewhich I was. Having the name, I grabbed the software and started looking at what it could do. Incredibly, the functionality I was looking for, which I thought would be the entirety of the software, was only one component in a sea of functionality.

And having run a BBS for years prior to going to college, the fact that I could install a PCI card and connect it to the phone network was not lost on me. After a couple of hours of looking at the software and getting it compiled, I started telling one of my teachers about the PCI cards and how maybe we could get some for the classroom for labs and such our classroom had 30 computers at 10 tables of 3.

He liked the idea and started talking to the program coordinator, and within about 30 minutes an order had been placed for 20 cards. Pretty amazing considering they were TDMPs decked out with four daughter cards, and they had only heard about them an hour prior to that. Then the obsession began. I spent every extra moment of that semester with a couple of computers dedicated to Asterisk use.

In those two months, I learned a lot. Then we had a co-op break. Luckily I had a lot of help on IRC for these were the days prior to any documentation on Asteriskand I learned a lot more during that semester. Seeing that the people who took a great interest in Asterisk at the time had a strong sense of community and wanted to contribute back, I wanted to do the same. Having no practical level of coding knowledge, I decided documentation would be something useful to start doing.

Besides, I had been writing a lot of papers at school, so I was getting better at it. One night I put up a website called The Asterisk Documentation Assigned TADA and started writing down any documentation I could. A couple of weeks later Jared Smith and I started talking, and started the Asterisk Documentation Project http: That project became the basis of the first edition of this book, Asterisk: From the first computer they purchased for far too much money when I was in grade 6 I started taking an interest in computers in grade 2 using a Commodore 64, and they got me a computer after a parent-teacher interview a few years later to letting me use the home phone line for my BBS endeavors and eventually getting me my own phone lineand everything else they have ever done for me, I can never thank them enough.

To my beautiful wife, Danielle, for setting the alarm every morning before she left for work, letting me sleep those extra 10 minutes before starting on this book, and understanding when I had to work late because I went past my 9 A.

There are so many people who help me and teach me new things every day, but the most influential on my life in Asterisk are the following: Mark Spencer for writing software that has given me a fantastic career, John Todd for his early examples, Brian K. West for his early help and enthusiasm on IRC, Steve Sokol and Olle Johansson for flying me to my first AstriCon and subsequent ones! When we set out to write the very first edition of this book over five years ago, we were confident that Asterisk was going to be a huge success.

Asterisk has proven that open source telecom is a lasting idea, and the open source telecom landscape is nowadays complemented by more than just Asterisk. I want to take this opportunity to thank my very good friend Leif Madsen, who has been with me through all three editions. I also want to thank Russell Bryant, who joined us for this edition, and whose dedication to this project and the Asterisk project in general is an inspiration to me.

To Jared Smith, who helped found the Asterisk Documentation Project and coauthored the first two editions with Leif and me but has since moved on to the Fedora projectI can only say: I would like to thank my business partners at Core Telecom Innovations and iConverged LLC, without whom I could not do all the cool things I get to do in my professional career. I would like to thank all my friends in the improv community, for helping me to keep laughing at all the challenges that life presents.

Finally, thanks to you, the Asterisk community. This book is our gift to you. I started working on Asterisk in I was a student at Clemson University and was working as a co-op engineer at ADTRAN in Huntsville, Alabama. My first job at ADTRAN was working in the Product Qualification department. I remember working with Keith Morgan to use Asterisk as a VoIP traffic generator for testing QoS across a router test network.

Meanwhile, a fellow co-op and friend, Adam Schreiber, introduced me to Mark Spencer. Over the next six months, I immersed myself in Asterisk. I learned as much as I could about Asterisk, telephony, and C programming. At the beginning ofI was hired by Digium to continue my work on Asterisk professionally. I have spent the past six amazing years working with Digium to improve Asterisk.

I have worked as a software developer, a software team lead, and now as the engineering manager of the Asterisk development team. I am extremely grateful for the opportunity to contribute to so many areas of the Asterisk project. There are many people that deserve thanks for the support they have provided along the way. To my wife, Julie, I cannot thank you enough for all the love and support you have given me. Thank you for keeping my life balanced and happy. You are the best.

To my parents, thank you for giving me so many great opportunities in my life to explore different things and find what I really enjoy. You taught me to work hard and never give up. To Leif and Jim, thank you for your invitation to contribute to this book. It has been a fun project, largely due to the pleasure of working with the two of you.

Thanks for the laughs and for your dedication to this book as a team effort. I have learned a lot from many people at Digium. There are three people who stand out the most as my mentors: Mark Spencer, Kevin P.

Fleming, and David Deaton. Thank you all for going the extra mile to teach me along the way. I am extremely grateful. To the software development team at Digium, thank you for being such an amazing team to work with. Your dedication and brilliance play a huge part in the success of Asterisk and make Digium a great place to work. To Travis Axtell, thank you for your help in my early days of learning about Linux and for being a good friend.

To my dogs, Chloe and Baxter, thanks for keeping me company while I worked on the book every morning. To the entire Asterisk community, thank you for using, enjoying, and contributing to Asterisk. We hope you enjoy the book! First they ignore you, then they laugh at you, then they fight youthen you win.

When we first set out—nearly five years ago—to write a book about Asterisk, we confidently predicted that Asterisk would fundamentally change the telecommunications industry.

Today, the revolution we predicted is all but complete. Asterisk is now the most successful private branch exchange PBX in the world, and is an accepted albeit perhaps not always loved technology in the telecom industry. Unfortunately, over the past five years the telecom industry has continued to lose its way. The methods by which we communicate have changed. Whereas 20 years ago phone calls were the preferred way to converse across distances, the current trend is to message via text email, IM, etc.

The phone call is seen as a bit of a dead thing, especially by up-and-coming generations. Asterisk remains pretty awesome technology, and we believe it is still one of the best hopes for any sort of sensible integration between telecom and all the other technologies businesses might want to interconnect with.

With Asterisk, no one is telling you how your phone system should work, or what technologies you are limited to. If you want it, you can have it. Asterisk lovingly embraces the concept of standards compliance, while also enjoying the freedom to develop its own innovations. What you choose to implement is up to you—Asterisk imposes no limits.

Naturally, this incredible flexibility comes with a price: Asterisk is not a simple system to configure. But when there are literally thousands of ways to achieve a result, the process naturally requires extra effort.

Perhaps it can be compared to building a house: Voice over IP VoIP is often thought of as little more than a method of obtaining free long-distance calling.

It is a simple goal, really, and it should be possible for us to make it happen in far more flexible and creative ways than are currently available to us. Technologies such as Asterisk lower the barriers to entry. When the Asterisk project was started inthere were other open-source telephony projects in existence. However, Asterisk, in combination with the Zapata Telephony Project, was able to provide public switched telephone network PSTN interfaces, which represented an important milestone in transitioning the software from something purely network-based to something more practical in the world of telecom at that time, which was PSTN-centric.

The Zapata Telephony Project was conceived of by Jim Dixon, a telecommunications consulting engineer who was inspired by the incredible advances in CPU speeds that the computer industry has now come to take for granted. Rather than having expensive components on the card, digital signal processing DSP [ 3 ] would be handled in the CPU by software.

Like so many visionaries, Dixon believed that many others would see this opportunity, and that he merely had to wait for someone else to create what to him was an obvious improvement. After a few years, he noticed that not only had no one created these cards, but it seemed unlikely that anyone was ever going to. At that point it was clear that if he wanted a revolution, he was going to have to start it himself.

And so the Zapata Telephony Project was born:. Since this concept was so revolutionary, and was certain to make a lot of waves in the industry, I decided on the Mexican revolutionary motif, and named the technology and organization after the famous Mexican revolutionary Emiliano Zapata.

Perhaps we should be calling ourselves Asteristas. Regardless, we owe Jim Dixon a debt of thanks, partly for thinking this up and partly for seeing it through, but mostly for giving the results of his efforts to the open source community.

Over the years, the Zapata Telephony interface in Asterisk has been modified and improved. Every PBX in existence suffers from shortcomings. No matter how fully featured it is, something will always be left out, because even the most feature-rich PBX will always fail to anticipate the creativity of the customer.

A small group of users will desire an odd little feature that the design team either did not think of or could not justify the cost of building, and, since the system is closed, the users will not be able to build it themselves. If the Internet had been thusly hampered by regulation and commercial interests, it is doubtful that it would have developed the wide acceptance it currently enjoys.

The openness of the Internet meant that anyone could afford to get involved. The tens of thousands of minds that collaborated on the creation of the Internet delivered something that no corporation ever could have. As with many other open source projects, such as Linux and so much of the critical software running the Internet, the development of Asterisk was fueled by the dreams of folks who knew that there had to be something more than what traditional industries were producing.

These people knew that if one could take the best parts of various PBXs and separate them into interconnecting components—akin to a boxful of LEGO bricks—one could begin to conceive of things that would not survive a traditional corporate risk-analysis process. While no one can seriously claim to have a complete picture of what this thing should look like, there is no shortage of opinions and ideas. Many people new to Asterisk see it as unfinished.

Perhaps these people can be likened to visitors to an art studio, looking to obtain a signed, numbered print. They often leave disappointed, because they discover that Asterisk is the blank canvas, the tubes of paint, the unused brushes waiting. Even at this early stage in its success, Asterisk is nurtured by a greater number of artists than any other PBX.

Most manufacturers dedicate no more than a few developers to any one product; Asterisk has scores. Most proprietary PBXs have a worldwide support team comprising a few dozen real experts; Asterisk has hundreds.

The depth and breadth of the expertise that surrounds this product is unmatched in the telecom industry. Asterisk enjoys the loving attention of old telco guys who remember when rotary dial mattered, enterprise telecom people who recall when voicemail was the hottest new technology, and data communications geeks and coders who helped build the Internet.

These people all share a common belief—that the telecommunications industry needs a proper revolution. Telecommunications companies that choose to ignore Asterisk do so at their peril. The flexibility it delivers creates possibilities that the best proprietary systems can scarcely dream of. Hackers built the networking engine that is the Internet. Hackers built the Apple Macintosh and the Unix operating system. Hackers are also building your next telecom system. Rather than being constricted by the dubious and easily cracked security of closed systems, the hackers will be able to quickly respond to changing trends in security and fine-tune the telephone system in response to both corporate policy and industry best practices.

Like other open source systems, Asterisk will be able to evolve into a far more secure platform than any proprietary system, not in spite of its hacker roots, but rather because of them. Never in the history of telecommunications has a system so suited to the needs of business been available, at any price. Asterisk is an enabling technology, and as with Linux, it will become increasingly rare to find an enterprise that is not running some version of Asterisk, in some capacity, somewhere in the network, solving a problem as only Asterisk can.

This acceptance is likely to happen much faster than it did with Linux, though, for several reasons:. Linux has already blazed the trail that led to open source acceptance. Asterisk is following that lead. The telecom industry is crippled, with no leadership being provided by the giant industry players. Asterisk has a compelling, realistic, and exciting vision. End users are fed up with incompatible and limited functionality, and horrible support. Asterisk solves the first two problems; entrepreneurs and the community are addressing the latter.

One of the compelling strengths of Asterisk is the passionate community that developed and supports it. This community, led by the fine folks at Digium, is keenly aware of the cultural significance of Asterisk and has an optimistic view of the future. The significance of this cooperation cannot be underestimated.

If the dream of Asterisk is to be realized, the community must continue to grow—yet one of the key challenges that the community currently faces is a rapid influx of new users.

Obviously, new users do not fit any particular kind of mold. While some will happily spend hours experimenting and reading various blogs describing the trials and tribulations of others, many people who have become enthusiastic about this technology are completely uninterested in such pursuits. Would one ask for a step-by-step guide to programming and expect to learn from it all that a language has to offer?

Asterisk is a different animal altogether, and it requires a totally different mind-set. As you explore the community, though, be aware that it includes people with many different skill sets and attitudes. As with any community, there are places where members of the Asterisk community meet to discuss matters of mutual interest. Of the mailing lists you will find at http: Anything commercial with respect to Asterisk belongs in this list.

If you want to buy an Asterisk service or product, post here. The Asterisk developers hang out here. The purpose of this list is the discussion of the development of the software that is Asterisk, and its participants vigorously defend that purpose. Expect a lot of heat if you post anything to this list not specifically relating to programming or development of the Asterisk code base.

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General coding questions such as queries on interfacing with AGI or AMI should be directed to the Asterisk-Users list. The Asterisk-Dev list is not second-level support! The Asterisk-Dev mailing list is about discussion of core Asterisk development, and questions about interfacing your external programs via AGI or AMI should be posted on the Asterisk-Users list.

This is where most Asterisk users hang out. This list generates several hundred messages per day and has over ten thousand subscribers.

While you can go here for help, you are expected to have done some reading on your own before you post a query. The Asterisk Wiki which exists in large part due to the tireless efforts of James Thompson — thanks James!

Another important resource is the community-maintained repository of VoIP knowledge at http: Since Asterisk documentation forms by far the bulk of the information on this website, [ 9 ] and it probably contains more Asterisk knowledge than all other sources put together with the exception of the mailing list archivesit is a popular place to go for Asterisk knowledge. An important new wiki project is the official Asterisk Wiki, found at http: While not yet as full of content as voip-info.

The Asterisk community maintains Internet Relay Chat IRC channels on irc. The two most active channels are asterisk and asterisk-dev. Over the past decade, in many cites around the world, lonely Asterisk users began to realize that there were other like-minded people in their towns.

Asterisk User Groups AUGs began to spring up all over the place. The Asterisk Documentation Project was started by Leif Madsen and Jared Smith, but several people in the community have contributed. The goal of the documentation project is to provide a structured repository of written work on Asterisk. In contrast with the flexible and ad hoc nature of the Wiki, the Docs project is passionate about building a more focused approach to various Asterisk-related subjects.

As part of the efforts of the Asterisk Docs project to make documentation available online, this book is available at the http: It is very rare to find businesses these days that do not have to reinvent themselves every few years. It is equally rare to find a business that can afford to replace its communications infrastructure each time it goes in a new direction. How compelling, then, to have a system that holds at its very heart the concept of openness and the value of continuous innovation.

So where to begin? Well, when it comes to Asterisk, there is far more to talk about than we can fit into one book. This book can only lay down the basics, but from this foundation you will be able to come to an understanding of the concept of Asterisk—and from that, who knows what you will build?

In a voice network, DSPs are primarily responsible for encoding, decoding, and transcoding audio information. This can require a lot of computational effort. Between the releases of Asterisk 1. Several projects have arisen to lower the barriers to entry for Asterisk. By far the most popular and well known is the FreePBX interface and the multitude of projects based on it. These interfaces check out http: Discussions related to programming external applications that interface with Asterisk via AGI or AMI are meant to be in asterisk.

Asterisk is very different from other, more traditional PBXs, in that the dialplan in Asterisk treats all incoming channels in essentially the same manner.

In a traditional PBX, there is a logical difference between stations telephone sets and trunks resources that connect to the outside world. Also, the concept of an off-site resource such as a reception desk is much more difficult to implement on a traditional PBX, because the system will not allow external resources any access to internal features. Asterisk, on the other hand, does not have an internal concept of trunks or stations. In Asterisk, everything that comes into or goes out of the system passes through a channel of some sort.

There are many different kinds of channels; however, the Asterisk dialplan handles all channels in a similar manner, which means that, for example, an internal user can exist on the end of an external trunk e. Unless you have worked with a traditional PBX, it may not be immediately obvious how powerful and liberating this is. Figure illustrates the differences between the two architectures. Asterisk is built on modules. We will discuss the use of many modules in this book. At this point we just want to introduce the concept of modules, and give you a feel for the types of modules that are available.

It is actually possible to start Asterisk without any modules at all, although in this state it will not be capable of doing anything. It is useful to understand the modular nature of Asterisk in order to appreciate the architecture. Call detail recording CDR modules. Channel event logging CEL modules. The details of how specific modules work will be covered in various chapters throughout the book, depending on what the module is and what it does. Some modules will be covered thoroughly; others may not be covered at all.

This module is ancient history. If you use it, be aware that you are mostly on your own when it comes to any sort of community support. This module is quite new, and its completeness and popularity are difficult to gauge at this time. Dialplan applications are used in extensions. The Dial application, for example, is responsible for making outgoing connections to external resources and is arguably the most important dialplan application.

Bridging modules are new in Asterisk 1. Each provides different features, which get used in different situations depending on what a bridge needs. You can store CDRs to a file defaulta database, RADIUS, or syslog. Call detail records are not intended to be used in complex billing applications. If you require more control over billing and call reporting, you will want to look at channel event logging, discussed next. The advantage of CDR is that it just works. Channel event logging provides much more powerful control over reporting of call activity.

By the same token, it requires more careful planning of your dialplan, and by no means will it work automatically. Without channel drivers, Asterisk would have no way to make calls.

Each channel driver is specific to the protocol or channel type it supports SIP, ISDN, etc. The channel module acts as a gateway to the Asterisk core. However, Zaptel is no longer supported by Asterisk and this driver has not been ported to DAHDI.

So if a call comes in on a PRI circuit using G. If a codec such as G. If you have a recording on a menu that has been stored as GSM, a format interpreter would need to be used to play that recording to any channels not using the GSM codec. If you store a recording in several formats such as WAV, GSM, etc.

They provide many useful enhancements to things like string handling, time and date wrangling, and ODBC connectivity. Forex trading income tax india PBX modules are peripheral modules that provide enhanced control and configuration mechanisms. Resource modules integrate Asterisk with external resources. Addon modules are community-developed modules with different usage or distribution rights from those of the main code.

They are kept in a separate directory and are not compiled and installed by default. To enable these modules, use the menuselect build configuration utility. When a problem does occur, it is very difficult for developers to solve unless they have a phone of the same model to test with. Test modules are used by the Asterisk development team to validate new code.

They are constantly changing and being added to, and are not useful unless you are developing Asterisk software. If you are an Asterisk developer, however, the Asterisk Test Suite may be of interest to you as you can build automated tests for Asterisk and submit those back to the project, which runs on several different operating systems and types of machines. By expanding the number of tests constantly, the Asterisk project avoids the creation of regressions in code.

By submitting your own tests to the project, you can feel more confident in future upgrades. More information about installing the Asterisk Test Suite is available in this blog post: Bse intraday tips for today information about building tests is available in this document: Asterisk is a complex system, composed of many resources.

These resources make use of the filesystem in several ways. Since Linux is so flexible in this regard, it is helpful to understand what data is being stored, so that you can understand where you are likely to find a particular bit of stored data such as voicemail messages or log files. The Asterisk configuration files include extensions. You will be working in this folder a lot as you configure and administer your Asterisk system. You will not normally have to interact with this folder; however, it will be occasionally useful to know where best ways to make money runescape p2p modules are located.

For example, if you upgrade Asterisk and select different modules during the menuselect phase of the install, the old incompatible modules from the previous Asterisk version will not be deleted, and you will get a warning from the install script.

Those old files will need to be deleted from the modules folder. There are several resources that require external data sources. System prompts also make money with your isp server to be stored somewhere on the hard drive. The spool is where Linux stores files that are going to change frequently, or will be processed by other processes at a later time. For example, under Linux print jobs and pending emails are normally written to the spool until they are processed.

For Asterisk, the spool is used to store transient items such as voice messages, call recordings, [ 16 ] how much money does machinist make files, and so forth.

When do you give earnest money on a short sale is capable of generating several different kinds of log files.

This folder will be extremely important for any troubleshooting efforts you undertake. The dialplan is the heart of Asterisk. All channels that arrive in the system will be passed through the dialplan, which contains the call-flow script that determines how the incoming calls are handled. Once you learn this language, it should be fairly easy to transition to AEL or LUA, should you desire.

Asterisk is capable of communicating with a vast number of different technologies. In general, these connections are made across a network connection; however, connections to more traditional telecom technologies, such as the PSTN, require specific hardware. Many companies produce this hardware, such as Digium the sponsor, owner, and primary developer of AsteriskSangoma, Rhino, OpenVox, Pika, Voicetronix, JunghannsDialogic, Xorcom, beroNet, and many others.

The authors prefer cards from Digium and Sangoma; however, the products offered by other Asterisk hardware manufacturers may be more suitable to your requirements. The most popular hardware for Asterisk is generally designed to work through the Digium Asterisk Hardware Device Interface known as DAHDI. These cards will all have different installation requirements and different file locations.

Handbook on forex trading nicholas tan will need to refer to the specific documentation provided by the manufacturers of any cards you install for details on those cards. The Asterisk release methodology has gone through a couple of iterations over the last few years, and this section is designed to help you understand what the version numbers mean.

Of particular relevance is the change in versioning that happened with the 1. When we had just Asterisk 1. Prior to the 1. Because all new development was done in trunk, until the 1. To try to relieve the pressure on administrators, and to enable faster access to new features in the time frame of months, and not yearsa new methodology was created. The goal was to provide new feature releases every 3—4 months which would be branched from trunk sports stock market oneseason, providing a shorter and clearer upgrade path for administrators.

So, so far we have branches, which are 1. Within each of those branches, we create tags releaseswhich look like 1. Unfortunately, it ended up not working out that 1.

Not only that, but the 1. People got confused as to what version to run, and that the 1. When you increase the number from 1. The development team learned a lot of things during the 1. The idea surrounding the releases was noble, but the implementation ended up being flawed when put into real use.

So, with Asterisk 1. While the development team still wants to provide access to new features and core changes on a more regular basis every 12 months being the goalthere is recognition that it is also good to provide long-term support to a stable, popular version of Asterisk. You can think of the Asterisk 1. The new LTS version is Asterisk 1. During the long-term support phase of Asterisk 1.

These will be tagged as versions 1. Each of these branches will receive bug fixes for a period of one year, and security releases will continue to be made for an additional year before the branches are marked as EOL. The current statuses of all Asterisk branches, their release dates, when they will go into security release—only mode, and when they will reach EOL status are all documented on the Asterisk wiki at https: Asterisk is composed of many different technologies, most of which are complicated in their own right.

As such, the understanding of Asterisk architecture can be overwhelming. Still, the reality is that Asterisk is well-designed for what it does and, in our opinion, has achieved remarkable balance between flexibility and complexity.

However, it is generally kludgy, limited in features, and requires complex, proprietary software to be installed in the PBX such as vendor-specific protocol extensions. WAV recordings will sound better and use less CPU. I long to accomplish great and noble tasks, but it is my chief duty to accomplish humble tasks as though they were great and noble. Pros and cons of stock option plans world is moved along, not only by the mighty shoves of its heroes, but also by the aggregate of the tiny pushes of each honest worker.

Many people shy away from this method, claiming that it is too difficult and time-consuming. Our goal here is to demonstrate that installing Asterisk from source is not actually that difficult to do. More importantly, we want to provide you with the best Asterisk platform on which to learn. In this book we will be helping quantitative stock market analysis build a functioning Asterisk system from scratch.

In this chapter you will build a $100 forex no deposit bonus 2014 platform for your Asterisk system. Given that we are installing from source, there is potentially a lot of variation in how you can do this.

As part of this process we will also explain installation of some of the software dependencies on the Linux platform that will be needed for topics covered later in this book such as database integration. We will show instructions for installing Asterisk on both CentOS a Red Hat—based distribution and Ubuntu a Debian-based distributionwhich we believe covers the vast majority of Linux distributions being installed today.

We have chosen to install on CentOS and Ubuntu because they are the most popular options, but Asterisk is generally distribution-agnostic. Asterisk will even install on Solaris, BSD, or OS X [ 18 ] if you like. There are also packages that exist for Asterisk that can be installed using popular package - management programs such as yum or apt-get.

You are welcome to experiment with them. Prebuilt packages may not always be kept up-to-date, though, so for the latest version we always recommend installing from source. You can find package instructions at http: Some commands you see in this chapter will be split into separate rows, each labeled for the distribution on which the command should be performed. Commands for which distributions are not specified are for common commands that can be run on both distributions.

Many projects have been created that use Asterisk as their underlying platform. Some of these, such as Trixbox, have become so popular that many people mistake them for the Asterisk product itself. These projects generally will take the base Asterisk product and add a web-based administration interface, a complex database, and a rigid set of constraints on how changes can be made to the configuration. We have chosen not to cover these projects in this book, for several reasons:.

This book tries, as much as possible, to focus on Asterisk and only Asterisk. Books have already been written about some of these Asterisk-based projects. We believe that if you learn Asterisk in the way that we will teach you, the knowledge will serve you well, regardless of whether you eventually choose to use one of these prepackaged versions of Asterisk.

For us, the power of Asterisk is that it does not attempt to solve your problems for you. These projects are an excellent example of what can be built with Asterisk.

They are truly amazing. However, if you are looking to build your own Asterisk application which is really what Asterisk is all aboutthese projects will impose limitations on you, because they are focused on simplifying the process of building a business PBX, not on making it easier to access the full potential of the Asterisk platform.

We recommend that you check them out. If you just want the nitty-gritty on how to get Asterisk up and running quickly, perform the following at the shell prompt. We encourage you to read through the entire chapter at least once, though, in order to better understand the full process.

Remember that Ubuntu requires commands to be prefixed with sudo. Perform a system update and reboot:. Synchronize time and install the NTP Network Time Protocol daemon:.

Some additional configuration of text files is required on Ubuntu. On CentOS, add a new system user:. For an Ubuntu install, we are assuming that the user created during the installation process is asteriskpbx. Get the latest source code via Subversion:. 99 binary options profit pipeline review additional sound prompts from menuselect:.

Modify the file permissions of the directories Asterisk was installed to:.

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Change the value of SELINUX from enforcing to disabledthen reboot. Copy the sample asterisk. Enable loading of modules automatically, and disable extra modules:. Save your changes and your module configuration is done. Your 99 binary options profit pipeline review is ready to configure your dialplan and channels.

Because Asterisk relies so heavily on having priority access to the CPU, it is essential that you install Asterisk onto a server without any graphical interface, such as the X Windowing system Gnome, KDE, etc. Both CentOS and Ubuntu ship a GUI-free distribution designed for server usage. We will cover instructions for both distributions. For more information about what CentOS is and its history, see http: You will need to download an World stock exchanges by market capitalisation from the CentOS website, located at http: You will then be presented with a list of mirrors that appear to be close to you physically.

Choose one of the mirrors, and you will be presented with a list of files to download. Likely you will want the first available selection, which is the first ISO file of a set. Upon booting from the CD, type linux text and then press Enter.

At this point the text installation interface will start. You will be asked whether you want to test the media. CentOS jquery get selected option value multiple then welcome you to the installation. Smart trader spreadsheet download Enter to continue.

Choose your language and make a keyboard selection. The installer will ask if you want to remove the existing partitioning scheme and create a new one. Select Remove all partitions on selected drives and create default layout.

If a more appropriate option exists, select that instead. In the drive window, verify that the correct disk drive is selected. Pressing Tab will cycle through the selections on the screen. Once the drive window is selected, you can scroll up and down presuming you have multiple drives and select which hard drive you wish to install to.

Toggle the selections by pressing space bar. Verify that the correct drive is selected, press Tab until the OK button is highlighted, and press Enter. A message confirming that you want to remove all Linux partitions and create the new partition scheme will be presented.

You will be asked to review the partitioning layout. Feel free to modify the partition scheme if you prefer something different see the following sidebar for some advice on this ; earn money fast simpsons tapped out, the default answer No is fine for light production use where storage requirements will be low. This is where Asterisk will store recordings, voicemails, log files, prompts, and a myriad of constantly growing information.

In normal operation, it is unlikely that Asterisk will fill the hard disk. However, if you have extensive logging turned on or are recording all calls, this could, in theory, occur. If the drive the operating system is mounted on fills up, there is the potential for a kernel panic. Having a full volume is still a major problem; however, you will at least be able to log into the system to rectify the situation. Selecting Yes will bring up the Partitioning tool. To partition the drive accurately, you need to know what the hard drive size is; this may not jibe with what is stamped on the outside of the drive because you have to tell day trading rules in nse tool how to chop up the drive.

Be sure the Activate on boot and Enable IPv4 support options are enabled, then select OK. If your binary option market maker system 3r provides automatic IP provisioning via DHCP, you can just select OK. Otherwise, select Manual address configurationenter the appropriate information, and then select OK.

You can either allow the DHCP server to provide one for you if your network assigns hostnames automatically or enter one manually, then select OK.

You will be presented with a list of time zones. Highlight your local time zone and select OK. How to make quick money in gta 3 this point, you will be asked for a root password. Enter a secure password and type it again to confirm. After entering your secure password, select OK. Next up will be the package selection.

Deselect all options in the list using the space barthen select the Customize software selection option. You will then be presented with the Package Group Selection screen. Scroll through the whole list, deselecting each item. A dependency check will then be performed and a confirmation that installation is ready to begin will be presented.

Select OK to start the installation. The filesystem will then be formatted, the installation image transferred to the hard drive, and installation of the system packages performed. Upon installation, you will be asked to reboot.

Remove any media in the drives and select the Reboot button. To do this, log in using the username root and the password you created during installation. Once logged in, run the following:.

When prompted to install the latest packages, press y and wait for the packages to update. When complete, reboot the system as it is likely the kernel will have been updated [ 25 ]:. This will remove all the bit libraries on your bit system, which can otherwise cause conflicts and issues when forex and treasury management course in mumbai Asterisk and other software.

Keeping accurate time is essential on your Asterisk system, both for maintaining accurate call detail records and for synchronization with your other programs. The ntpd command can be used to ensure that the time on your Asterisk server remains in sync with the rest of the world:.

The Ubuntu server install process asks you to add a system user other than root, but CentOS does not. Now we need to provide the asteriskpbx user sudo access. We do this by modifying the sudoers file with the visudo command. With the sudo -related applications and file installed, we can modify the sudoers file. Execute the visudo command and look for the lines shown below:. Modify it like so:. Save the file, log out from root by typing exitand log in as the asteriskpbx user you created.

Test your sudo access by running the following command:. One last thing needs to done, which will allow you to enter stock market hours fourth of july without having to enter the full path.

Start by opening the hidden file. As previously, save the file by pressing Esc and then typing: Ubuntu Server is a popular Linux distribution loosely based on Debian. There is also a popular desktop version of the software. The Ubuntu Server package contains no GUI and is ideal for Asterisk installations. To get the latest version of Ubuntu Server, [ 27 ] visit http: You will be provided with a page that contains information about Ubuntu Server Edition.

Clicking the orange Download button in the upper-right corner will take you to a page where you can select either the bit or bit version of Ubuntu Server. After selecting one of the options, you can press the Start download button. Upon booting from the CD, you will be presented with a screen where you can select your language of choice.

By default English is the selected language, and after a timeout period, it will be automatically selected. After selecting your language, press Enter. The next screen will give you several options, the first of which is Install Ubuntu Server. Select it by pressing Enter. You will then be asked which language to use for the installation yes, this is slightly redundant.

Select your language of choice the default is Englishand press Enter. You will be presented with a list of countries. You will then be asked if you would like to use the keyboard layout detector. If you know which keyboard type you have, you can select No and then pick it from a list of formats. If you are utilizing the keyboard layout detector, you will be prompted to press a series of keys.

If you use the keyboard detector and it does not detect your keyboard correctly typical when installing into a virtual machine via a remote consoleyou can go back and select from a list manually. If all goes well, you will be prompted to enter a hostname for your system. You can pick anything you want here, unless your network requires your system to a have a specific hostname. Input it now and then press Enter. The installer will attempt to contact a Network Time Protocol NTP server to synchronize your clock.

Ubuntu will then try to autodetect your time zone and present you with its choice. Select your time zone, or select from the worldwide list if your time zone is not shown. The installer will then ask you some questions about partitioning your system. Typically the default is fine, which is to use the guided system, utilizing the entire disk, and to set up the Logical Volume Manager LVM.

Select Yes and press Enter to continue. You will now be asked how much space to use the default value will be to use the entire disk. The installer will then request one last confirmation before making the changes to the disk. Select Yes to write the changes to disk. The installer binary options methods 123 templates hackers now format the hard disk, write the partitioning scheme to disk, copy the files, and perform the file installation.

The system will suggest a username, but you are free to change the username to whatever you like. Ubuntu does a good job of providing a secure system by not providing direct access to root, but rather using the sudo application, which allows you to run commands as root without being the root user.

Enter a username, [ 28 ] such as asteriskpbxand a secure password to continue. The installer will then ask you if you want to encrypt your home directory.

This is not necessary and will add CPU overhead. The rest of the installation instructions will assume that asteriskpbx was chosen as the username. If your system is behind a web proxy, enter the proxy information now. You will then be asked if you want to install updates automatically. The default is to perform no automatic updates, which is what we recommend. It is better practice to identify updates on a regular basis and perform them manually in a controlled manner.

Normally, you would want to advise your users of the expected downtime and schedule the downtime to happen after business hours or while a redundant system is running. Select No automatic updates and press Enter. SSH is essential if you wish to perform remote work on the system. However, if your local policy states that your server needs to be managed directly, you may not want to install the OpenSSH server. Pressing the Enter key will accept the current selections and move on with the install.

You need to use space bar to toggle your selections. If this is the only operating system on the machine which it likely isUbuntu will give you the option to install the GRUB bootloader on your system. It provides this prompt in order to give you the option of skipping the GRUB installation, as it will modify the master boot record MBR on your system. If this is the only operating system installed on your server, select Yes. Once logged in, run the following command:.

The password that sudo wants is the password you just logged in with. Press Enter when prompted to continue, at which point the latest package updates will be installed. When complete, reboot the system for the changes to take effect as the kernel has probably been updated. Keeping accurate time is essential on your Asterisk system, both for maintaining accurate call detail records as well as for synchronization with your other programs.

The default on Ubuntu is to run a time sync server without ever changing the time on your own machine. Because of this, we need to guide you through using a command line editor. The nano editor is already installed on your Ubuntu machine and is remarkably easy to use [ 30 ]:. Use your arrow keys to move down to the section that looks like. Add two new lines after this section, to allow ntpd to synchronize your time with the crystal dashboard stock market world, such that the above section now looks like.

When prompted whether to save the modifications, press Y ; nano will additionally ask you for the filename. The next section provides an in-depth review of the installation process.

These packages will allow you to build a basic Asterisk system, along with DAHDI and LibPRI. Not every module will be available at compile time with these dependencies; only the most commonly used modules will be built.

If additional dependencies are required for other modules used later in the book, instructions will be provided as necessary. Please be aware that the dependency information on CentOS bit does not take into account that bit libraries should not be installed. If such libraries are installed, you will end up with additional packages that use disk space and can cause conflicts if the system attempts to compile against a bit library instead of its bit counterpart. In order to resolve this problem, add.

So, for example, instead of executing yum install ncurses-develyou will execute yum install ncurses-devel. This is not necessary on a bit platform. Software dependencies for Asterisk on Ubuntu Server and CentOS Server. Note that you will also require the software dependencies for each package that we indicate needs to be installed. These will be resolved automatically for you when you use either yum or apt-get. We have also included the OpenSSL development libraries, which are not strictly necessary to compile Asterisk, but are good to have: We have installed vim as our editor, but you can choose anything you want, such as nanojoeor emacs.

Asterisk contains a script that will install the dependencies for all features in Asterisk. At this time it is complete for Ubuntu but does not list all required packages for CentOS. For certain software dependencies, a third-party repository may be necessary. This appears to be most often the case when using CentOS. A couple of repositories that seem to be able to provide all the extra dependencies required are RPMforge http: We may occasionally refer to these third-party repositories when they are required to obtain a dependency for a module we are trying how to make money in runescape low level build and use.

There are several methods of getting Asterisk: Typically, package-management systems will have versions that are older than those available from Subversion or the downloads site, and we want to make sure we have the most recent fixes available to us, so we tend to avoid them. The official packages from Digium do tend to stay up to date.

Once everything is built, it will be installed with the sudo command. To begin, issue the following command:. Choose one of the following two methods to get your files:. Asterisk is a constantly evolving project, and there are many different versions of the software that you can implement.

The concept of how Asterisk is versioned is important to understand because the versioning system for Asterisk has undergone a few changes of methodology over the years.

Having said all that, in most cases all you need to do is grab the latest version from the http: We will be installing crude oil etf options using Asterisk 1. Subversion is a version control system that is used by developers to track changes to code over a period of time. Each time the code is modified, it must first be checked out of the repository; then it must be checked back in, at which point the changes are logged.

Thus, if a change creates a regression, the developers can go back to that change and remove it if necessary. This is a powerful and robust system for development work. It also happens to be useful for Asterisk administrators seeking to retrieve the software. To download the source code to the latest version of Asterisk 1. The preceding commands will retrieve the latest changes to the source in that particular branch, which are changes that have been made after producers livestock market san angelo latest release.

If you would prefer to use a released version, please refer to the next section. To obtain the latest released versions of DAHDI, LibPRI, and Asterisk using the wget application, issue the following commands:. The next step is to compile and install the software, so onward to the next section. With the source files downloaded you can compile the software and install it. The order for installing is:.

LibPRI [ 31 ]. DAHDI [ 32 ]. Asterisk [ 33 ]. Installing in this order ensures that any dependencies for DAHDI and Asterisk are installed prior to running the configuration scripts, which will subsequently ensure that any modules dependent on LibPRI or DAHDI will be built. LibPRI is a library that adds support for ISDN PRI and BRI.

Check out the latest version of LibPRI and compile it like so:. You can also download the source via wget from http: The Digium Asterisk Hardware Device Interfaceor DAHDI formerly known as Zaptelis the software Asterisk uses to interface with telephony hardware. DAHDI is actually a combination of two separate code bases: The forex 50 pips system from rita numbering for DAHDI-linux-complete will look something like 2.

There are also FreeBSD drivers for DAHDI, which are maintained by the community. These drivers are available at http: Another dependency is required for installing DAHDI, and that is the kernel source. It is important that the kernel version being used match exactly that of the kernel source being installed.

You can use uname -a to verify the currently running kernel version:. The following commands show how to install DAHDI-linux-complete 2. There may be a newer version available by the time you are reading this, so check downloads. Binary options methods 123 templates hackers there is a newer version available, just replace the version number in the commands:.

You will need to have Internet access when running the make all command, as it will attempt to download the latest hardware firmware from the Digium servers. After installing DAHDI, we can move on to installing Asterisk. With both DAHDI and LibPRI installed, we can now install Asterisk:.

With the files now installed in their default locations, we need to modify the permissions of the directories and their contents. There is an additional step that is not strictly required, but is quite common and arguably important: Execute the following commands after running make install which we did previously:.

Change the last line of the dahdi. With that out of the way, we can move on to performing the base configuration that should be done after all installations. These are the steps that all system administrators will need to start out with when installing a new system. If the commands that need to be run differ on CentOS and Ubuntu, you will see a table with rows labeled for each distribution; otherwise, you will see a single command that should be run regardless of which Linux distribution you have chosen.

In CentOS, the Security-Enhanced Linux SELinux system is enabled by default, and it often gets in the way of Asterisk. Sometimes the issues are quite subtle, and at least one of the authors has spent a good number of hours debugging issues in Asterisk that turned out to be resolved by disabling SELinux. You can reenable SELinux by doing the same thing, but replacing the 0 with a Change the SELINUX option from enforcing to disabled.

Alternatively, you can managed mini forex accounts reviews the value of enforcing to permissivewhich profitunity trading system logs the errors instead of enforcing the policy.

In order to get Asterisk up and running cleanly, we need to create some configuration files. We could potentially install the sample files that come with Asterisk by executing the make samples command in our Asterisk source and then modify those files to suit our needs, but the make samples command installs many sample files, most of them for modules that you will never use. Running make samples on a system that already has configuration files will overwrite the existing files.

Using make samples to Create Sample Configuration Files for Future Reference. Even though we are not going to use the sample configuration files that come with Asterisk, the fact is that they are an excellent reference. If there is a module that you are not currently using but wish to put into production, the sample file will show you exactly what syntax to give the money binary options trading free, and what options are available for that module.

Running the sudo make samples command in your Asterisk source directory [ 35 ] is harmless on a new system that has just been built, but it is very dangerous to run on a system that already has configuration files, as this command will overwrite any existing files which would be a disaster for you if you do not have a current backup. The first file needed is indications. This is done with the asterisk. Near the old school runescape best way to make money f2p of the [options] section, there are two options we need belasting over stock options enable: Uncomment the runuser and rungroup lines, and modify them so that they each contain asteriskpbx as the assigned value.

Then modify the file by uncommenting the two lines starting with runuser and rungroup and modifying the value to asteriskpbx. We now have all the configuration files required to start a very minimal version of Asterisk.

Asterisk will start successfully without any errors or warnings although it does warn you that some files are missingand present to you the Asterisk command-line interface CLI. At this point there are no modules, minimal core functionality, and no channel modules with which to communicate, but Asterisk is up and running.

Executing the module show command at the Asterisk CLI shows that there are no external modules loaded:. Asterisk can be run either as a daemon or as an application. In general, you will want to run it as an application when you are building, testing, and troubleshooting, and as a daemon when you put it into production.

However, without any arguments, this command will assume certain defaults and start Asterisk as a background application. In other words, you never want to run the command asterisk on its own, but rather will want to pass some options to it to better define the behavior you are looking for. The following list provides some examples of common usages. Gt binary options command displays a helpful list of the options you can use.

For a complete list of all the options and their descriptions, run the command man asterisk. This option starts Asterisk as an application in the foreground. This means that Asterisk is tied to your user session.

In other words, if you close your user session by logging out or losing the connection, Asterisk dies. This is the option you will typically use when building, testing, and debugging, but you would not want to use buying apple shares in the uk option in production.

If you started Asterisk in this manner, type core stop now at the CLI prompt to stop Asterisk and exit. This option can be used with other options e.

It does exactly the same thing as the CLI command core set verbose n stock broker vancouver bc n is any integer between 0 and 5 any integer greater than 5 will work, but will not provide any more verbosity. For example, if you are looking to see only startup errors, notices, and warnings, leaving verbosity off will prevent all the other startup messages from being displayed. This option can be used in the same way as -vbut instead of normal output, this will specify the level of debug output which is primarily useful for developers who wish to troubleshoot problems with the code.

You will also need to enable output of debugging information in the logger. This command is essential if you want to connect to the CLI of an Asterisk process running as a daemon. You will probably use this option more than any other for Asterisk systems that are in production. This option will only work if you have a daemonized instance of Asterisk already running. To exit the CLI when this option has been used, type exit.

This command allows you to pass a string to Asterisk that will be executed as if it had been typed at the CLI. We recommend you try out a few combinations of these commands to see what they do. Restarts Asterisk automatically after a crash. Can be configured to email the administrator if a crash has occurred.

Executes a script if a crash has occurred. In most environments this script works fine in its default format.

Create the file modules. Using cat to Quickly Create Files and Add Content to Them. There are many cases in a Linux system where it is necessary to create a file, and then add some content to it. This is commonly done by using the touch command to create the file, and then opening it with an editor to add the content.

However, there is a less-known way of doing this that lets you create the file and add the content all at once:. Paste or type the content you want to add to the file. If you wanted to, you could leave the file like this, and Asterisk would simply load any modules it found in the modules folder. With your new modules. You can verify this by starting Asterisk and running the module show command:. We now have many modules loaded, and many additional dialplan applications and functions at our disposal.

There are, of course, other modules that you could remove, and others that you may find extremely useful, so feel free to tweak this file as you wish. Ideally, you should be loading only the modules that you need for the system you are running.

The examples in this book assume that your modules. Additional information about the modules. By defining different classes, you can specify different hold music to be used in various situations, such as different announcements to be played while holding in a queue, or different hold music if you have multiple PBXs hosted on the same system. Additional information about musiconhold. The modules are what give Asterisk its power and functionality. New modules are constantly being created.

In the installation sections, we conveniently skipped over using the menuselect system in order to keep the instructions simple and straightforward. However, it is important enough that we have given menuselect its own section. However, the following example will give you an idea of how menuselect works, and is recommend for any installation. By default Asterisk only installs the core sound prompt files, and only in GSM format.

Also, the three OpSound music on hold files available for download are only selected in. We can do this with the menuselect system by running make menuselect in the Asterisk source directory. There are two interfaces available for menuselect: The solution is to use the newt-based menuselect system.

To get the newt-based menuselect working, you need to have the libnewt development libraries installed:. You can do this with the following commands:. Run the following commands to start menuselect:.

You can use the arrow keys on your keyboard to move up and down. The right arrow key will take you into a submenu, and the left arrow key will take you back. You can use the space bar or Enter key to select and deselect modules. Pressing the q key will quit without saving, while the x key will save your selections and then quit. Whenever you install a dependency, you will always need to rerun configure before you run menuselectso that the new dependency will be properly located.

The dependant module will at that point be available in menuselect. If the module selection still contains XXXeither the configure script is still unable to find the dependency or not all dependencies have been satisfied. You will be presented with a list of available options. These options represent the core sound files in various languages and formats. By default, the only set of files selected is CORE-SOUNDS-EN-GSMwhich is the English-language Core Sounds package in GSM format.

The reason we have multiple formats for the same files is that Asterisk can play back the appropriate format depending on which codec is negotiated by an endpoint. This can lower the CPU load on a system significantly. After selecting the appropriate sound files, press the left arrow key to go back to the main menu. Then scroll down two lines to the Extra Sound Packages menu and press the right arrow key or Enter.

You will notice that by default there are no packages selected. As with the core sound files, select the appropriate language and format to be installed. A good option is probably to install the English sound files in the WAVULAWand ALAW formats. You then need to install your new prompts by downloading them from the Asterisk downloads site. This is done simply by running make install again:. Your Asterisk server will need to have a working Internet connection in order to retrieve the files.

Administrators often build tools when performing installations on several machines, and Asterisk is no exception.

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If you need to install Asterisk onto several machines, you may wish to build a set of scripts to help automate this process. The menuselect system contains command-line options that you can use to enable or disable the modules that are built and installed by Asterisk. If you are starting with a fresh checkout of Asterisk, you must first execute the configure script in order to determine what dependencies are installed on the system.

Then you need to build the menuselect application and run the make menuselect-tree command to build the initial tree structure:. You will be returned output like the following:.

The options displayed can then be used to control which modules are installed via the menuselect application. If you then look at the menuselect. Enabling those modules can be done with the following command, but before doing that look at the current menuselect. After executing the following command, look at the menuselect.

The modules listed in the menuselect. You can then build the menuselect. When we say updating your system, that is quite different from upgrading your system.

Updating your system is the process of installing new minor versions of the same branch. For example, if your system is running Asterisk 1. In contrast, we use the term upgrade to refer to changes between Asterisk branches major version number increases. So, for example, an upgrade would be going from Asterisk 1. You should be able to install new files directly into the appropriate directories.

Upon installation, however, you may get a message like the following:. This most often occurs when you have installed modules in one version of Asterisk, and then installed a new version of Asterisk without compiling those modules as the installation process will overwrite any modules that existed previously, replacing them with their upgraded versions.

There is a caveat here, though: It is recommended that you keep a directory with your third-party modules in it that you can reinstall from upon update of your Asterisk system. Downloading third-party modules into this directory allows you to easily reinstall those modules when you upgrade.

Be sure to change the file permissions to match those of the user running Asterisk! Any time you install additional packages, you will need to run the.

This message means you have not installed the wget application, which is required for you to download packages from the Asterisk downloads site, for Asterisk to download sound files, or for DAHDI to download firmware for hardware. This means that the Asterisk configure script is unable to find your C compiler, which typically means you have not yet installed one. Be sure to install the gcc package for your system. This error is encountered when you have not installed the make application, which is required to build Asterisk.

You will encounter this error when the XML parser libraries are not installed. These are required by Asterisk 1. You will get this error when attempting to build DAHDI without having installed the Linux headers, which are required for building Linux drivers.

Upgrading Asterisk between major versions, such as from 1. Once a phone switch is in production, it is terribly disruptive for that system to be unavailable for nearly any length of time, and the upgrade of that phone system needs to be well thought-out, planned, and tested as much as possible prior to deployment.

And because every deployment is different, it is difficult, if not impossible, for us to walk you through a real system upgrade. However, we can certainly point you in the right direction for the information you require in order to perform such an upgrade, thereby giving you the tools you need to be successful. A production Asterisk system should never be upgraded between major versions without first deploying it into a development environment where the existing configuration files can be tested and reviewed against new features and syntax changes between versions.

For example, it may be that your dialplan relies on a deprecated command and should be updated to use a new command that contains more functionality, has a better code base, and will be updated on a more regular basis. Commands that are deprecated are typically left in the code for backward-compatibility, but issues reported about these deprecated commands will be given lower priority than issues to do with the newer preferred methods.

There exist two files that should be read prior to any system upgrade: These files contain details on changes to syntax and other things to be aware of when upgrading between major versions. The files are broken into different sections that reference things such as dialplan syntax changes, channel driver syntax changes, functionality changes, and deprecation of functionality, with suggestions that you update your configuration files to use the new methods.

Another thing to consider when performing an upgrade is whether you really need to perform the upgrade in the first place. An alternative to upgrading the entire system is simply to add functionality to your system by running two versions simultaneously on separate systems. By running separate boxes, you can access the functionality added to a later version of Asterisk without having to disrupt your existing production system.

You can then perform the migration more gradually, rather than doing a complete system upgrade instantly. Two parts of Asterisk should be thoroughly tested when performing an upgrade between major versions: These two parts of Asterisk rely on testing your code to make sure any cleanup of syntax changes in either the AMI or the AGI, or added functionality, does not interfere with your existing code.

By performing a code audit on what your program is expecting to send or receive against what actually happens, you can save yourself a headache down the road. The testing of call detail records CDRs is also quite important, especially if they are relied upon for billing. The entire CDR structure is really designed for simple call flows, but it is often employed in complex call flows, and when someone reports an issue to the tracker and it is fixed, it can sometimes have an effect on others who are relying on the same functionality for different purposes.

Upgrading Asterisk can be a successful endeavor as long as sufficient planning and testing are carried out prior to the full rollout. In this chapter we looked at how to install an operating system one of Ubuntu or CentOS and Asterisk itself. We did this securely by installing via sudo and running Asterisk as the non-root user asteriskpbx. We are well on our way to building a functional Asterisk system that will serve us well.

In the following chapters we will explore how to connect devices to our Asterisk system in order to start placing calls internally and how to connect Asterisk to outside services in order to place phone calls to endpoints connected to the PSTN and accept calls from those endpoints. Nevertheless, this chapter may make the process look complex. This is simply because we have an obligation to ensure you are provided with all the information you need to accomplish a successful install.

A system with a 1 terabyte drive can store somewhere in the range of 2 million minutes of telephony-quality recordings. DAHDI is a set of Linux kernel modules used with Asterisk. The nano editor has been selected for its ease of use and its handy on-screen instructions. We even know a developer at Digium who uses it while writing code for Asterisk, though most people tend to use more complex editors such as emacs or vim. However, we are going to install it for the sake of completeness.

It is also required for the MeetMe conferencing application. Again, we will install this for completeness. There are only three songs installed by default, and callers will quickly tire of listening to the same three songs over and over again. All you need to know here is that outside of North America and Japan, A-law is used. In the last chapter, we covered how to install Asterisk. But where should you get started with configuration? That is the question this chapter answers. There are a few common configuration files that are relevant regardless of what you are using Asterisk to accomplish.

In some cases they may not require any modification, but you need to be aware of them. There is a sample asterisk. Asterisk will look for asterisk. To specify a different location for asterisk. For most installations of Asterisk, changing the directories is not necessary. However, this can be useful for running more than one instance of Asterisk at the same time, or if you would like files stored in nonstandard locations.

This section of the asterisk. Most of these are also controllable via command -line parameters to the asterisk application. For a complete list of the command -line options that relate to these options, see the Asterisk manpage:. An alternative is to run keys init at the Asterisk command line. The result may be that transcoding is required for a call that would not normally require it. This section of asterisk. It is primarily used by remote consoles asterisk -r. Occasionally the Asterisk development team decides that the best way forward involves making a change that is not backward-compatible.

Although most modules do not use much in the way of resources, and they all load very quickly, it just seems cleaner to our minds to load only those modules that you are planning on using in your system. Additionally, there are security benefits to not loading modules that accept connections over a network. In the past we felt that explicitly loading each desired module was the best way to handle this, but we have since found that this practice creates extra work.

After every upgrade we found ourselves having to edit the modules. What we prefer to do these days is to allow Asterisk to automatically load the modules that it finds, but to explicitly tell Asterisk not to load any modules we do not want loaded by use of the noload directive.

Using menuselect to Control Which Modules Are Compiled and Installed. One other way that you can control which modules Asterisk loads is to simply not compile and install them in the first place. During the Asterisk installation process, the make menuselect command provides you with a menu interface that allows you to specify many different directives to the compiler, including which modules to compile and install.

With the exception of autoloadall of the options may be specified more than once. The sounds that people expect from the telephone network vary in different parts of the world.

Different countries or regions present different sounds for events such as dialtone, busy signal, ringback, congestion, and so forth. In the early days of Asterisk this file only contained sounds for a limited number of countries, but it is now quite comprehensive.

To assign the tones common for your region to channels, you can simply assign the tonezone using the CHANNEL function, and that tonezone will apply for the duration of the call unless changed later:. However, since signaling from a call could come from various places from the carrier, from Asterisk, or even from the set itselfyou should note that simply setting the tonezone in your dialplan does not guarantee that those tones will be presented in all situations.

If you have too much time on your hands, you can do all sorts of pointless but entertaining things with your indications. For example, fans of Star Wars can make the following change to the end of their indications.

If you then use the country named ' starwars ' in your configuration files or dialplan, any ringing you pass back will sound quite different from the standard ring you are used to.

Try the following dialplan code to test out your new ringing sound:. Depending on the type of device used to call into this example, you may wonder if it will actually work. SIP phones, for example, typically generate their own tones instead of having Asterisk generate them.

This example was carefully crafted to ensure that Asterisk will generate a ringback tone to the caller. The key is the Answer that is executed first. While Asterisk will run without an indications. DAHDI does not use the indications.

If your system supports multiple countries for example, if you have a centralized Asterisk system that has users from different regionsyou may not be able to simply define the default country. In this case, you have a couple of options:. Define the country in the channel definition file for the user. Define the country in the dialplan using the CHANNEL tonezone function. Part of the problem with music on hold is that while in the past it was common to just plug a radio or CD player into the phone system, the legal reality is that most music licenses do not actually allow you to do this.

So how to deal with this? There are two legal ways: What we will do, however, is show you how to take the music you have and make it work with Asterisk. There are several websites that offer music that has been released under Creative Commons or other licenses. Each song may have its own licensing requirements, and just because you can download a song for free does not mean you have permission to use it as music on hold.

Be aware of the licensing terms for the music you are planning to use for your music on hold. While Asterisk can use MP3s as a music source, this method is not at all ideal. MP3s are heavily compressed, and in order to play them the CPU has to do some serious work to decompress them in real time. This is fine when you are only playing one song and want to save space on your iPod, but for music on hold, the proper thing to do is convert the MP3 to a format that is easier on the CPU.

Since CentOS does not have MP3 capability installed with soxyou will have to install mpg before you can convert MP3 files for use with Asterisk. First you will need to install the rpmforge repository. To find out which version you need, open your web browser and go to http: You need to make sure this new repository is used correctly, so run the following:. If you want to know more about yum priorities, see this site: Once the repository has been added, you can proceed to get mpg If you are familiar with the file formats and have some experience working with audio engineering software such as Audacity, you can convert the files on your PC and upload them to Asterisk.

To convert your MP3 files to a format that Asterisk understands, you need to run the commands outlined here in this example we are using a file named SilentCity. Then, downsample the resulting WAV file to a sampling rate that Asterisk understands:. If you have not done so already, install soxand the libsox-fmt-all package:.

Then, convert your MP3 file directly to the uncompressed SLN format:. In newer versions of sox e. You now need to reload musiconhold in Asterisk in order to have it recognize your new files:. To test that your music is working correctly, add the following to the [UserServices] context in your dialplan:.

Dialing from one of your sets should play a random file from your moh directory. This chapter helped you complete some initial configuration of Asterisk. From here you can move on to setting up some phones and taking advantage of the many features Asterisk has to offer. Configuring a channel in Asterisk for the device to connect through is relatively straightforward, but you also need to configure the device itself so it knows where to send its calls.

SIP is a peer-to-peer protocol, and while it is common to have a setup where endpoints act as clients and some sort of gateway acts as a server, the protocol still thinks in terms of peer-to-peer relationships. What this means is that a SIP telephone expects to make a direct connection to another SIP telephone, without a PBX in between.

The reality is that many SIP transactions happen through a server, and in the case of Asterisk, it is common to have the PBX in the middle of all connections. When a SIP call is made from a telephone to another telephone through Asterisk, there are actually two calls happening: Asterisk bridges the two channels together.

From the perspective of the SIP telephone, therefore, you need to configure it to send all its calls to Asterisk, even though the device is quite capable of directly connecting to another SIP endpoint without the Asterisk server.

The SIP protocol is complex and very flexible, and configuring endpoints can seem difficult because they have much more flexibility than we require of them for an Asterisk implementation. The set will identify itself and download customized files that define the required parameters for that telephone. As an example, these could be XML files on an FTP server. The exact download process and syntax of these files will differ from manufacturer to manufacturer. In this chapter we will only talk about the configuration of sets from the perspective of Asterisk.

Before we get started with configuring Asterisk for our telephones, we are going to recommend some best practices regarding telephone naming, abstracting the concepts of users, extension numbers, and telephones from each other. In Asterisk, all the system cares about is the channel name. There is really no concept of a user at all, [ 43 ] and extensions are simply ways of directing call flow through the system.

For example, your dialplan might inform Asterisk that when extension number is requested it should call the phone on my desk, but extension could just as easily call a company voicemail box, play back a prompt, or join a conference room. We can even specify that extension should ring the device on my desk from Monday to Friday between 9 A.

Inversely, when a call is made from a device during business hours, the callerID could show a daytime number, and the rest of the time could show an after-hours number many reception desks become security desks at night.

The concept of an extension in Asterisk is crucial. In most PBXs, an extension is a number that you dial to cause a phone or service to ring. In Asterisk, an extension is the name of a grouping of instructions in the dialplan.

Yes, an Asterisk extension could be a number such as that rings a phone, but it could just as easily be a name such as voicemail that runs a sequence of dialplan applications. The abstraction between the name of an extension and what that extension does is a powerful concept in Asterisk, as extension could do a number of things depending on any number of variables that are programmed into the system.

This is especially relevant in the context of features such as hot-desking. Hot-desking is a feature that allows someone to log into a device and receive his calls at that device.

Since they are unlikely to be on-site at the same time, instead of having a separate telephone for each of those three sales agents, they could share a single office phone or on a larger scale, a dozen folks could share a pool of, say, three phones.

This scenario illustrates the convenience and necessity of allowing the system to separate the concept of a user and extension from the physical phone. So what are some examples of bad names for telephone devices? The same reasoning can be applied to why you would not want to name a phone based on an extension number: A popular way to name a phone is using the MAC address of the device. The choice is yours as to how you want to name your phones, but we primarily want to abstract any concept of the telephone being owned by a person, or even its location in the network, since these concepts are outside the realm of Asterisk and can change at any time.

You will want to use phone names that match the hardware you are using or some other string that is unique to the device you are registering. As a final consideration, we should make it clear that what we are suggesting regarding device names is not a technical requirement. There are three types of endpoints you would typically provide your users with that could serve as a telephone set. They are popularly referred to as hardphones, softphones, and Analog Terminal Adaptors ATAs.

A hardphone is a physical device. It looks just like an office telephone: A softphone is a software application that runs on a laptop or desktop. More recently, softphone applications have been written for smart phones that allow you to connect to other networks other than just the cellular network. The interface of the softphone is often styled to look like a physical telephone, but this is not necessary. An ATA is designed to allow traditional analog telephones and other analog devices, such as fax machines, cordless phones, paging amplifiers, and such to connect to a SIP network, [ 44 ] and will typically be a sandwich-sized box that contains an RJ connector for the phone commonly referred to as an FXS portan RJ connector for the network, and a power connector.

Some ATAs may support more than one phone. Hardphones have the advantage that the handsets have good acoustic properties for voice communications. Any decent-quality telephone is engineered to pick up the frequencies of the human voice, filter out unwanted background noise, and normalize the resulting waveform. People have been using telephones for as long as the telephone network has existed, and we tend to like what is familiar, so having a device that communicates with Asterisk using a familiar interface will be attractive to many users.

Also, a hardphone does not require your computer to be running all the time. Disadvantages to hardphones include that they are nonportable and expensive, relative to the many quality softphones on the market today that are available for free.

Also, the extra clutter on your desk may not be desirable if you have limited work space, and if you move around a lot and are not generally at the same location, a hardphone is not likely to suit your needs although, one at each location you frequent might be a valid solution.

Softphones solve the portability issue by being installed on a device that is likely already moving with you, such as your laptop or smart phone. Because many softphones are free, it is likely that the first telephone set you connect to Asterisk will be a softphone. Also, because softphones are just software, they are easy to install and upgrade, and they commonly have other features that utilize other peripherals, like a webcam for video calling, or perhaps an ability to load files from your desktop for faxing.

Some of the disadvantages of softphones are the not-always-on nature of the devices, the necessity to put on a headset each time you take a call, and the fact that many PCs will at random times during the day choose to do something other than what the user wants them to do, which might cause the softphone to stop working while some background task hogs the CPU. ATAs have the advantage of allowing you to connect to your SIP network analog devices, [ 45 ] such as cordless phones which are still superior in many cases to more advanced types of wireless phones [ 46 ]paging amplifiers, and ringers.

ATAs can also sometimes be used to connect to old wiring, where a network connection might not function correctly. The main disadvantage of an ATA is that you will not get the same features through an analog line as you would from a SIP telephone. This is technology that is over a century old.

Asterisk will happily allow you to interact with the outside world in ways that were scarcely dreamed of only a few years ago. As we see more unification of communications applications with the popularity of social networks, communities such as Skype, and more focus on network-based services such as those provided by Google, the flexibility and popularity of software-based endpoints will continue to grow.

The blurring of the lines between voice and applications is constantly evolving, and softphones are well positioned to rapidly respond to these changes. Asterisk allows devices using many different protocols to speak to it and therefore to each other.

However, the SIP and IAX2 protocols are the most popular and mature VoIP modules, so we will focus our attention on them. The configuration for the other protocols is similar, and the sample configuration files are full of information and examples, so once you have the basics down, other protocols should be fairly easy to work with. The channel configuration files, such as sip.

Common information about the channel driver is contained at the top of the configuration file, in the [general] section. All section names are encased in square brackets, including device names. Anything that follows a section name or device definition, which for our purposes is essentially the same thing is applied to that section. The [general] section can also contain information to define defaults for device configurations, which are overridden in the section for each device, or in a template.

Asterisk also comes with defaults that are hardcoded, so while some settings are mandatory, many other settings can be ignored as long as you are happy with the defaults.

Asterisk will check for parameters in the following order:. Check the specific section for the relevant channel. Check the template for the section.

Check the [general] section. Use the hardcoded defaults. If you are not sure, set the parameter explicitly in the section of the configuration file that deals with that specific channel, or in the relevant template. The dialplan is the heart of an Asterisk system: Both the relevant channel configuration file and the extensions.

When a call comes into Asterisk, the identity of the incoming call is matched in the channel configuration file for the protocol in use e. The channel configuration file also handles authentication and defines where that channel will enter the dialplan. Once Asterisk has determined how to handle the channel, it will pass call control to the correct context in the dialplan. The context parameter in the channel configuration file tells the channel where it will enter the dialplan which contains all the information about how to handle and route the call.

Conversely, if the dialplan has been programmed to dial another device when the request for extension number is being processed, a request to dial telephony device FFFF will use the channel configuration file to determine how to pass the call back out of the dialplan to the telephone on the network including such details as authentication, codec, and so forth.

A key point to remember is that the channel configuration files control not only how calls enter the system, but also how they leave the system.

So, for example, if one set calls another set, the channel configuration file is used not only to pass the call through to the dialplan, but also to direct the call from the dialplan to the destination. The SIP [ 47 ] channel module is arguably the most mature and feature-rich of all the channel modules in Asterisk.

If you look through the sip. Fortunately, the default options are normally all you need, and therefore you can create a very simple configuration file that will allow most standard SIP telephones to connect with Asterisk. This is a standard section that appears at the top of the configuration file for all channel modules, and must always be named in this way. The [general] section contains general configuration options for how that protocol relates to your system, and can be used to define default parameters as well.

You should be aware that we could have used any name we wanted, and also that there needs to be an identically named context in extensions. Keep in mind that for some channels you may actually want to accept unauthenticated calls. A common use for allowing unauthenticated calls is for companies that allow dialing by uniform resource identifiers URIslike email addresses.

If we wanted to allow customers to call us from their phones without having to authenticate, we could enable guest calls and handle them in the unauthenticated context defined by the previous option.

You may be wondering why you might ever want to allow unauthenticated calls. The reason is that if you publish your SIP URI on your business cards e. What you want instead is for your unauthenticated context to put incoming calls into a controlled environment. The srvlookup option is used to enable Asterisk to perform a lookup via a DNS SRV record, which is typically used for outbound connections to service providers.

The udpbindaddr [ 49 ] option takes the value of an IP address or 0. Alternatively, we could limit VoIP connections for this protocol to a single interface by defining the IP address of a specific network interface on our system. Currently in Asterisk the udpbindaddr and tcpbindaddr options are an all-or-one proposition.

As of version 1. As an example, consider the different values for the udpbindaddr option:. The tcpenable option allows us to accept requests via the TCP network transport protocol. There are also tlsenable and tlsbindaddr options for enabling SIP over TLS encrypted SIP.

Following the section name with! By doing this we eliminate the need to repetitively add and change configuration options for every device we choose to define.

If you want to change something for an individual device that was previously defined in the template for that device, you can do that under the section header, and it will override what was defined by the template. It is not necessary to use templates, but they are extremely handy, and we use them extensively. This tells the channel driver to attempt to match on name first, and then IP address. There are two other type definitions you can use: The difference between them has to do with how Asterisk interprets incoming SIP requests.

The rules are covered in this table: When a request from a telephone is received and authenticated by Asterisk, the requested extension number is handled by the dialplan in the context defined in the device configuration; in our case, the context named LocalSets. The host option is used when we need to send a request to the telephone such as when we want to call someone.