Introduction to Asterisk - The multiprotocol Open Source PBX

An introduction to Asterisk for a beginner - from installation to configuration and your first phone call.

John Todd, CTO, VOIP, Inc

John Todd has been involved with the Internet industry since 1994, as a member of the founding teams of Digex, Cidera, and other notable Internet infrastructure providers. His principal expertise is in the area of provisioning system development, deployment, and integration, in environments ranging from co-location to leased-line to satellite to GIS. He is also an accomplished project manager in the areas of software development, provisioning, network operations, and customer service, having managed large groups of technical staff through multiple layers of management hierarchy. His current interests are focused in the areas of Voice-over-IP systems integration, operations, and large-scale provisioning. He has run an active consultancy, advising more than 20 companies on future voice technology, and currently serves as the Chief Technology Officer for VOIP Inc., which is a holding company for several hardware, software, and services product lines all involved with Voice over IP. John is a 1993 graduate of the University of Rochester, and a regular participant in operations and technology conferences like NANOG and BSDCon.

Creating the dialplan: Tips and tricks

The Asterisk dialplan, extensions.conf, is the heart of your Open Source PBX installation. It's in the dial plan you configure various functions, like voicemail-on-no-answer, call forwarding, call pickup, transfers and queue handling.

This tutorial gives you the basics of how to construct a dial plan and delivers some useful examples that you can implement in your own installation.

• Basics: Extensions, priorities and contexts
• Interactions with other configuration files
• Variables
• Conditional expressions
• Macros
• Example configurations

Brian Capouch, Palaver.net

Brian Capouch is the Chair of the Computer Science Department at Saint Joseph's College in Rensselaer, Indiana. He is also a partner in Peoples Wireless, an ISP that services three counties in Northwestern Indiana. He was one of Vonage's first customers, and is currently heavily involved in a variety of VoIP activities, including using VoIP on his fixed wireless network. He has a Master's degree in Agricultural Systems Management from Purdue University, and lives on a small farm near Monon, Indiana.
Brian recently purchased an old railroad hotel built in 1853, the oldest standing structure in Pulaski County, Indiana. He hopes to restore it to its original condition.

Connecting Asterisk to the PSTN - a tutorial

Asterisk supports a number of connections to a PSTN network, being able to work as an ordinary PSTN PBX or as a multiprotocol VOIP/PSTN PBX and gateway.

This tutorial covers how to connect Asterisk to the telephony network, PSTN, over ISDN links as well as analogue telephone lines.

• E1/T1 PRI
• TDM Cards
• ISDN BRI

Paul Mahler, founder and Chief Technical Officer, Signate

Paul is the author of the book "VoIP Telephony with Asterisk", the first comprehensive guide to Asterisk. A specialist for three decades in the architecture and implementation of mission-critical business systems, Mahler's experience encompasses telephony, relational databases, and UML and model driven architectures. His clients have included Air Touch, AT&T, Pacific Bell, Verizon and Visa. He began his career as the founder of Horizon Software systems, the first software applications developer for the UNIX operating system.

Mahler has a BA, at honors, from the University of California at Berkeley in Philosophy of Science. He completed four years in the Ph.D. program of Bio-physics at U.C. Berkeley and the graduate program of Medical InformationSciences at U.C. San Francisco.

Visualising Asterisk: various user- or administator-friendly interfaces

Asterisk by default is managed with a command-line character based user interface in a Unix system. In a modern world, one would require more. Asterisk has a management interface, that can be used to build both user-oriented, management-oriented and receptionist-oriented user interfaces. This session is an overview of several user interfaces and a summary.

Featured Software:

• AstGUIclient: This presentation will cover the issues involved with creating a system that allows up to 100 GUI clients to connect to and interact with an Asterisk server through the Asterisk manager interface. Also to be covered are the issues involved with multi-server Asterisk environments and the special needs of corporate and call-center applications.

Target audience:

• Asterisk administrators
• Developers that works with user interfaces

Moderator: James H. Thompson, Commpartners
(Maintainer of the wiki at voip-info.org)

Speakers:

• James H. Thompson: Overview and introduction
• Matt Florell, IT Director, VICI Marketing: The AstGUIclient (Open Source)
• Jason Penton: A Flash based Asterisk User Interface
• Senad Jordanovic

Matt Florell has over eight years experience in building and managing corporate IT systems specializing in Customer Relationship Management(CRM) systems, databases, scalable applications and system optimization. Matt is currently the IT director of VICI Marketing in Clearwater Florida and is the creator and maintainer of the astGUIclient project.

Jason Penton is currently reading for his PhD in computer science at Rhodes University, Grahamstown, South Africa. He has been studying in the area of VoIP and telecoms for the past 4 and a half years, focusing on VoIP protocols, H.323, SIP and MGCP. He has also been working for Telkom S.A (the national telecoms carrier in S.A. since 2001.

Implementing CLASS features with Asterisk

This presentation offers an overview of how to implement Vertical Service Codes (VSCs) using the Asterisk application programmers interface (API). VSCs are subscriber-dialed “*XX” codes used to access CLASS features typically offered by local exchange service providers or enterprise PBXs, including call forwarding, speed dialing, callback, last number recall, privacy calling, etc. The presentation shows that, by using the Asterisk API, VSCs can be available across multiple channel technologies, which affords rapid prototyping of new features using Perl scripting via the AGI. The VSCs are made available by simple modifications to the “extensions.conf” file. Per-user feature control can be provided using an abstracted database interface. The presentation concludes with an example using a SIP subscriber base.

This presentation is ideally suited for two different audience types:

• The Asterisk development community as a framework for adding CLASS features.
• The user community wanting to offer CLASS features to an existing subscriber base.

Jerry D. Doty

NSP/NCM Software Engineering Director
sentitO Networks, Inc

Introduction to Call queues and agents

A tutorial on how to implement Asterisk in a call center environment. The tutorial covers the strength of Asterisk for call centers and the different features that can be developed to support this business.

• Choice of softphones
• Screen Pop for Inbound calling
• Dialer that allows predictive, progressive, preview and broadcast dialing
• Full digital recording and Screen capture for all agents interaction
• Remote management application (web enabled)
• Detailed reporting and statistics

François Lambert, COO, Atelka

François Lambert has over 12 years of experience in management, design, analysis and implementation of interactive business solutions. Companies such Microcell, Air Canada, Ontario Lottery and others profited from M. Lambert's experience in the call center technology. In addition, François masters the design and the development of complex applications for major call centers. His skills include C, C++, VXML, JAVA, CTI, Telephony, IVR and CRM applications.

Supporting Asterisk

If you deliver Asterisk in-a-box solutions or asterisk services to customers you will end up supporting Asterisk the way Digium support their customers.
In this tutorial, you will get an insight of how to track down and debug common problems on the various zaptel interfaces and VoIP protocols, how to use the bug tracker (Mantis), and a more advanced section on how to debug deadlocks and other asterisk issues using gdb.

Matthew Fredrickson, Digium Support

Matthew Fredrickson started with Digium about 3 years ago where he was first exposed to Asterisk. He has been involved in various development and support activies while working there. His primary activities at Digium include answering your phone calls :-) , writing patches, and doing general development on Asterisk. He is also currently working to receive his undergraduate degree in Computer Engineering at the University of Alabama in Huntsville.

Asterisk on FreeBSD: Call with the Daemon

What's the easiest way to get Asterisk up and running on FreeBSD? How does one configure FreeBSD specific features such as boot scripts, security, logging and reporting? How does one tune a FreeBSD system to best support VOIP?

• Installing and configuring Asterisk on FreeBSD.
• Comparison of Asterisk on FreeBSD and Linux.
• Asterisk management on FreeBSD.
• FreeBSD firewall configuration for SIP and IAX2.
• VOIP quality of service bandwidth management on FreeBSD.
• Configuring NAT and NATed Asterisk.

You will learn the path of lowest resistance to getting Asterisk up and running on FreeBSD, including preferred components and configurations. You will also learn how to leverage best-of-breed security and management features of FreeBSD.

Rich Murphey, CEO, White Oak Labs

Dr. Murphey holds a Ph.D. in Electrical and Computer Engineering from Rice University and has fifteen years of experience in software development as an author, architect, product line manager, a business development liaison and board member. He specializes in rapid execution for emerging markets.
Dr. Murphey is best known for open-source contributions as a co-founder of the FreeBSD Unix operating system and the Xfree86 Project window system. He served for 12 years on each project on both the board of directors and the core development team.
Rich currently serves on the boards of the XFree86 Project, Inc. and Infragard Houston. He is a member of the ACM, IEEE, WISA, ISSA, SANS and CIS. He is a frequent speaker at national conferences, most recently on technical advancements in intrusion prevention software. His main interests are operating systems and Intrusion Detection Systems.

Designing a IP network for VOIP services and more

Service providers offering VOIP services must seriously take into account the design of their own, and other peoples, IP networks.

This tutorial will dive into the dirty details of designing and building out a VOIP service offering. At the end the participant will have a solid understanding of the following issues and how to potentially address them:

• Core IP network design and routing
• Measuring and Monitoring network performance
• IP Traffic Engineering, planning for growth
• PSTN Traffic Engineering, planing for growth
• Difference between transit and peering, how to select the right mix
• "Stupid IP Routing Tricks" to help create a redundant network
• Design and deployment of DNS servers
• How to use NAPTR, SVR, TXT and other DNS resource records to make life
  easier
• ENUM and E.164
• Dynamic DNS and ENUM
• Semi-Self documenting systems (CVS, RANCID, et al)
• NAT
• IPv6 directions
• 911 services and connectivity
• Network security, both IP and PSTN

John Brown, co-founder and CEO, Chagres Technologies, Inc.

Founded in 2001, Chagres provides consulting services for organizations that wish to design, deploy and or expand their TCP/IP based networks. Chagres Technologies, thru Mr. Brown provided the principal network engineering for IANA's L Root DNS Server. Mr. Brown is a past member of the ARIN Advisory Council, is an active participant in NANOG (North American Network Operators Group), the IETF, RIPE, APNIC, and ARIN regional IP registries. Chagres has helped several organizations design and deploy highly available global networks, including VOIP service
offerings. Mr. Brown has been actively designing and deploying sucessful TCP/IP based networks since 1986. He has been active in the creation and operation of several Internet Exchange Points, or IX's.
Chagres Technologies, is a privately held company with offices in Albuquerque, Rotterdam, and Sydney.

Asterisk - building your system for performance and scalability

.... from a small embedded pc to a large load balanced failover smp opteron cluster.... choose the most suited codec and hardware for your needs.

• How many simultaneous calls can your hardware handle for your setup?
• Codec and jitter buffer choices
• Simulating calls to test the performance of your setup
• Tweaking your OS for larger Asterisk setups
• Basic load balancing and failover for asterisk servers + drawbacks
• How to combine a SIP Proxy, SER, with Asterisk

Joachim Vanheuverzwijn (Zoa)

Joachim Vanheuverzwijn is a 25 year old part time expatriot in Bulgaria, working for securax ltd where he is providing voip consultancy to a 300 seat in- and outbound callcenter (partially running on Asterisk and gnugk). In Belgium, he's doing consultancy for a Belgian telecom provider.
When he's not scratching his 'Bulgarian radioactive mosquito'-bites he's probably being chased by wild dogs or enjoying the aftermath of another homebrew...

Advanced SIP tutorial: routing constructs in SIP

The SIP protocol provides a very powerful set of routing constructs that can be used to provide load balancing, high availability, location services and build higher level applications. This tutorial explains the use of the SIP Record-Route capability as well as how DNS SRV, ENUM can be applied to SIP. The tutorial will cover how SIP enables a more distributed architecture not possible with traditional centralized switch based protocols. Readily available open source implementations of these building blocks will be explored. Additionally some open source applications of these constructs will be discussed.

Alan Hawrylyshen, CTO and Chief Architect
Jasomi Networks Inc.

Over the past decade, Alan Hawrylyshen has held responsibility for large scale, life- and mission- critical software systems with real time performance requirements. Prior to joining Jasomi, Hawrylyshen was CTO of Polyphase Design, which provided software designs and implementations for the Canadian Air Traffic Control System. His team also provided integration services for other programs associated with air traffic control systems.
Mr. Hawrylyshen is a significant contributor to the SIP open source community with implementations of the STUN protocol and as a founding developer on the reSIProcate stack project. Mr. Hawrylyshen is currently the reSIProcate project administrator (www.resiprocate.org).

Advanced Asterisk Configuration Wizardry

Two Asterisk gurus and contributors take you on a wild tour into the dark corners of Asterisk Wizardry. You will learn more on how to automate the configuration process with dynamic databases, how to set up dynamic conferences and other advanced Asterisk topics.

This tutorial is aimed at Asterisk administrators with some knowledge on how to configure, install and run Asterisk.

• Asterisk extensions.conf preprocessor
• Call Queue example
• Call Parking example
• Loading configs from SQL (res_config)
  • Why use res_config?
  • Advantages/Disadvantages
  • # include's and usage with res_config
• Dynamic extensions.conf
  • Usage of Macros to create dynamic configs
  • #includes
• Advanced tips and tricks
• Q&A

Brian K. West (bkw_) and Josh Roberson (twisted)