pre-conference sessions ...docs.digium.com/astricon/astricon2014-guide.pdf · this year we find...
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WELCOME . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
AGENDA AT A GLANCE . . . . . . . . . . . . . . . . . . . . . . . . . 2 - 3
HELPFUL INFO . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 - 5
HOTEL MAP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 - 7
PRE-CONFERENCE SESSIONS . . . . . . . . . . . . . . . . . . 9 - 11
KEYNOTES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12 & 18
CONFERENCE SESSIONS . . . . . . . . . . . . . . . . . . . . . . . 9 - 22
SPEAKER BIOS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19 - 31
EXPO FLOORPLAN . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
ASTRICON SPONSORS . . . . . . . . . . . . . . . . . . . . . . . . . 33 - 35
ALL THINGS ASTERISK . . . . . . . . . . . . . . . . . . . . . . . . . 36 - 37
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October 22-24, 2014 | Las Vegas
Thank you to our2014 AstriCon Sponsors!
This year we find ourselves in Las Vegas, such a great venue, for an AstriCon that promises to be our best yet!
It’s been a whole year and much has happened in the world of Asterisk... the hard work invested in Asterisk 12 last year by the the dev team and our valued Community Members has been built on and matured by the same people to bring us Asterisk 13, an LTS (long term stability) release, which, as I write this message, is available as a beta2 release.
What strikes me most about this AstriCon is the vast array of excellent conference sessions we have for you this year, and although I ‘curate’ this central element of the event, it is the Asterisk Community members that have submitted these sessions that must really take the credit for the diversity, depth and quality of the 2014 crop!
Rarely can an event boast such a breadth of related subject matter, such a mighty collection of exactly the right people to ‘talk Asterisk’ or such an interesting cross-section of the ecosystem populating the exhibit floor - and I would like you to join me in saying a big ‘Thank you!’ to all of our sponsors.
Matt Jordan (head of the Asterisk dev team) will be running a couple of sessions to allow you to catch up with what has gone into Asterisk 13 and how to make most effective use of it through the ARI (the Asterisk REST Interface), a new API that arrived in Asterisk 12 and has been improved in Asterisk 13.
With more than 45 hand-picked sessions to choose from, there will inevitably times when you wish you could be in two (or possibly three or four) places at once! This is the beauty of attending AstriCon with your colleagues, in that it gives you the ability to cover a variety of sessions - and then explain them to each other afterwards. If that’s not possible, don’t fret, as we do record the sessions - so you will be able to catch up with what you have missed after the event.
I wanted to highlight a couple that have an extra special flavor this year... First, two AstriCon favorites, the lovely Allison Smith (The Voice of Asterisk) and Ben Klang (Mojo Lingo) are joining forces to present a talk called “Using Asterisk to create ‘Her’” which will delve into integrating speech technologies with Asterisk, and even flirt with the idea of adding artificial intelligence into the mix. Also, we are delighted to welcome Jim ‘Dude’ Dixon back to AstriCon. It was Jim that started to low cost telephony card revolution with the Zapata Telephony Project and the Tormenta card. One of our sessions, “¡Viva la Revolución!”, will be an audience with Jim, where he will recount what led him to the powerful conclusions he came to back in the day, and he may even share his thoughts on where things are going next!
AstriCon is such a great place to learn a huge amount about Asterisk, what people are doing with it - and how they’re doing it, in addition to hearing about all sorts of associated projects and wild and wacky integrations... But do not attend AstriCon simply to receive, please be prepared to share your thoughts and opinions, ask your questions and think about the talk that YOU might give next year!
You can also participate in the ‘Fastest Dude’ contest. This year we’re going back to basics... It’s just you, a SIP phone and Asterisk... If you can rise to the challenge and be one of the three ‘Fastest Dudes’ a sweet prize will be yours! Look for the contest booth on the exhibition floor.
I hope you will enjoy the event and find the value planted in every single session.
Once again, welcome to AstriCon - the conference and exhibition that YOU make the Asterisk event of the year!!
Worldwide Community Director, Asterisk
October 22, 2014
Greetings fellow Asterisk enthusiasts and welcome to AstriCon 2014!
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WEDNESDAY, OCTOBER 22, 2014 — PRE-CONFERENCE DAY
9:00am - 5:00pm
Asterisk from Scratch (pg. 11) Summerlin F
Digium Commercial Sessions (pg. 9)Charleston A
Distro Sessions (pg. 11) Veranda C
Hackathon (pg. 11 & 17) Summerlin A
Melissa Shepherd, Asterisk Training Manager, DigiumJustin Hester, Asterisk Technical Trainer, Digium
Session 1 (9:00am - 10:30am)Intro to Asterisk & Asterisk Architecture
Session 2 (11:00am - 12:15pm)Installing Asterisk & Intro to Dialplan
Session 3 (1:45pm - 3:00pm)Configuring Endpoints & Interactive Dialplan
Session 4 (3:30pm - 4:55pm)Advanced Dialplan
9:00am - 9:35am: Web Commmunications for Business, Steve Sokol, Respoke
9:35am - 10:35pm: Phone & GatewayAPI - Doug Bailey, Digium; Diego Gosmar, xCALLY; Gus Trague, NEXACOMM
11:00am - 11:30am: Switchvox Anywhere - Deployment Options, Adam Kramer
11:30am - 12:05pm: Education Panel - Clark County, Casper College
1:45pm - 2:20pm: Digium Cloud Services, Mark Amick
2:20pm - 2:55pm: Building Your Brand with Switchvox, Brian Ferguson
3:30pm - 4:00pm: Partnering with Digium, Jim Butler
11:00am - 11:30am: Using the FreePBX EndPoint Manager, Luke Duquaine, Schmooze Com Inc.
11:30am - 12:05pm: Elastix Call Center PRO, Elvita Crespo, Elastix
1:45pm - 2:20pm: FreePBX Phone Apps for Digium Phones Presentation, Tony Lewis, Schmooze Com Inc.
2:20pm - 2:55pm: Elastix 3.0 Jorge Gutierrez, Elastix
8:00am - 9:00am: Check-in
9:00am - 5:00pm: Hacking
5:00pm - 7:00pm EXPO Hall Reception & Fastest Dudes Contest, Summerlin B Hackathon Judging, Summerlin A
THURSDAY, OCTOBER 23, 2014 — CONFERENCE DAY 1 9:00am - 9:45am KEYNOTE - Craig Walker, CEO and Founder, Uberconference --- Charleston B (pg. 12)
Developers & Tutorials - Summerlin A (pg. 13) Business & Case Studies - Charleston A (pg. 12) Security & Asterisk-Based Distros - Veranda DE (pg. 15) Service Provider & Contact Center - Summerlin F (pg. 16)
10:00am - 10:35am Testing with SIPPClod Patry
Building a Conference Bridge SystemMartin Kaplan
Making your Asterisk System SecureEric Klein
System Monitoring Enterprise Asterisk ClustersJames Pearson
11:00am - 11:35am Building Rick Communications Applications Using ARIMatt Jordan and Samual Fortier-Galaneau
Maintaining an International Multi-Site Asterisk NetworkCassius Smith
Updates in Fraud Protection and PreventionFlavio E. Goncalves
Getting Started with WebRTCBilly Chia
11:40am - 12:15pmScaling FastAGI Applications with Go
Lefteris ZafirisAsterisk at the Heart of Interactive Media
Sebastian CoutureVoice and Security: What’s happening in the Real World
Erik KnightContact Center Scaling with Asterisk
Corey McFadden
1:45pm - 2:20pm Twelve Reasons Your API SucksKeith Casey
Marketing Boot CampSusan Elder
VoIP Security: Common Mistakes, Prevention Methods, and Unforseen Attack Vectors
Phillip Mullis
Fronting Asterisk with Kamailio for WebRTC and Web-service IntegrationPeter Dunkley
2:25pm - 3:00pm Distribution, Redundancy, and High Availability Using OpenSIPSAli Pey
The New Rules for IVRAllison Smith
Security Panel Chris Brown, Flavio E. Goncalves, Eric Klein, Phillip Mullis
Automatic Configuration Management for Kamailio and Asterisk in the era of Puppet
Giacomo Vacca
3:30pm - 4:05pm Getting the Best Out of WebRTCTim Panton and Dan Jenkins
Telecommunications Abiertas de Mexico: How we increased Safety for Journalists & Human Rights Defenders
Juan Carlos Fernadez
FreePBX Yesterday, Today and TomorrowPhilippe Lindheimer
Kamailio - SIP Servers EverywhereDaniel Constantin-Mierla
8:00pm - 11:00pm All Conference Party --- Cherry Lounge
FRIDAY, OCTOBER 24, 2014 — CONFERENCE DAY 2 9:00am - 9:45am KEYNOTE - Russell Bryant, Principal Software Engineer, Red Hat --- Charleston B (pg. 18)
Developers & Tutorials - Summerlin A (pg. 19) Business & Case Studies - Charleston A (pg. 20) Security & Asterisk-Based Distros - Veranda DE (pg. 21) Service Provider & Contact Center - Summerlin F (pg. 22)
10:00am - 10:35am WebRTC: The Big Debate, Shut UP, and Build SomethingLantre Barr
FreePBX High Availability Tony Lewis
What’s in Asterisk 13?Matt Jordan
Tipping the ScalesBen Klang
11:00am - 11:35amIntroducing Payload-Voice: Call Queues for the Asterisk REST
InterfacePaul Berlanger
Cloud-Based Asterisk - Tips from the Front LineNir Simonovich
¡Viva la Revolución! Jim “Dude” Dixon
Interfacing with Asterized Internet of Things NodeMichael Schloh von Bennewitz
11:40am - 12:15pm Connecting Non-SIP IP Camera to Your PBXTom de Moor
Asterisk Collaboration on Web-Functions and Adapt to Carrier - Grade Specifications
Takahiko Nagata
Using Asterisk to create “Her”Ben Klang and Allison Smith
Integrating Asterisk with a Feature-Rich BrowserPaul Ashfield
1:45pm - 2:20pmDangerous Demos
James Body
How to Sell it Better: A Case of WebRTC in RetailJakub Klausa
Open SIPs for Asterisk UsersPeter Kelly
Developing an IVR Payment System with AsteriskKayode Akintunde
2:25pm - 3:00pm Creating ARI apps FAST!Matt Riddell
Asterisk, Elastix, and FreePBX ClinicScott Griepentrog and Mark Michelson; Elvita Crespo and Jorge
Gutierres; Tony Lewis and Andrew Nagy
Service Provision using Asterisk and OpenSIPSBogdan-Andrei Iancu
3:00pm - 3:30pm AstriCon Roadmap Session --- Charleston B
4:00pm - 6:00pm AstriCon Wrap Party --- Cherry Lounge
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Hours-at-a-GlanceWednesday, October 22nd Thursday, October 23rd Friday, October 24th
Fastest Dude to the Dialtone Contest (full details on page 8)
5:30pm - 7:00pm Exhibit Hall - Summerlin B
3:30pm - 4:45pm Exhibit Hall - Summerlin B
EXPO/Exhibit Hall5:00pm - 7:00pm
Summerlin B10:35am - 5:00pm
Summerlin B10:35am - 2:00pm
Summerlin B
Keynotes9:00am - 9:45am
Craig WalkerCharleston B
9:00am - 9:45am Russell BryantCharleston B
Conference Sessions
9:00am - 5:00pm Distro. Track: Veranda C
Comm. Track: Charleston AAsterisk from Scratch:
Summerlin B
10:00am - 4:05pm Summerlin ACharleston AVeranda DESummerlin F
10:00am - 3:30pm Summerlin ACharleston AVeranda DESummerlin F
dCAP Testing (by appointment only)
8:30am - 11:30amWillows
8:30am - 11:30amWillows
8:30am - 11:30amWillows
Parties and Networking
EXPO Opening Reception 5:00 pm - 7:00 pm
Summerlin BAll-Conference Party 8:00pm - 11:00pm
Cherry Lounge (must show your AstriCon
Conference Badge to attend)
Wrap Party BBQ4:00 - 6:00pm Cherry Lounge
(must show your AstriCon Conference Badge to attend)
dCAP reception - invitation only5:00pm - 6:00pm
180 Suite, Red Rock Resort
Registration, Breakfast, Lunch and Breaks ScheduleWednesday, October 22 Thursday, October 23 Friday, October 24
Registration Open7:30am - 7:00pm
Outside Summerlin A7:30am - 5:00pm
Outside Summerlin A7:30am - 3:00pm
Outside Summerlin A
Continental Breakfast7:30am - 9:00am
Summerlin/Charleston hallway7:30am - 9:00am
Summerlin/Charleston hallway7:30am - 9:00am
Summerlin/Charleston hallway
Morning Coffee Break10:35am - 11:00am
Summerlin/Charleston hallway10:35am - 11:00am
Summerlin B10:35am - 11:00am
Summerlin B
Lunch12:15pm - 1:45pm
Pool Area12:15pm - 1:45pm
Pool Area12:15pm - 1:45pm
Pool Area
Afternoon Coffee Break3:00pm - 3:30pm
Summerlin/Charleston hallway3:00pm - 3:30pm
Summerlin B
*The Hotel is an entertainment/casino business governed by the laws of the State of Nevada and licensed by the State of Nevada Gaming Commission. It is against the law for any minor (under the age of 21) to loiter in or near any gaming area. It is also against policy for minors (under the age of 21) to be unattended in the hotel rooms. Please be advised that an adult (21 years or older) must be present at the time of check-in and throughout the entire stay.
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STAY CONNECTED WITH EVERYTHING
ASTERISK & ASTRICON
WEB:www.asterisk.org • www.astricon.net
BLOG:blogs.digium.com
FACEBOOK:www.facebook.com/astriconpage#astricon
TWITTER:@asteriskpbx | @astricon | @digium#astricon
INSTAGRAM@digium, #digium, #astricon
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Fastest Dude to the Dialtone ContestStep up and flex your Asterisk muscles! Once again we have a challenge that could have even the most seasoned professional scratching their heads. You know the score… It’s the fastest dudes that collect the prizes. We don’t want to give too much away right now, but just know that the more experienced you are with Asterisk basic con-figuration files the better!
Contest Times: (located in Exhibit Hall)
October 22nd: Contest from 5:30pm - 7:00pm
October 23rd: Contest from 3:30pm - 4:45pm
Prizes will be awarded in Friday’s Keynote session.
PRIZES:1st: GoPro HERO 3+ camera2nd: Apple iPAD mini 16GB
3rd: Beats Solo HD On-ear headphones
E m p o w e r i n g C o m m u n i c a t i o n
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Digium Commercial Sessions - Charleston A
Web Communications for Business 9:00am - 9:35am Steve Sokol, Director of Strategic Programs, Digium The web communications revolution is underway! New technologies, like WebRTC and WebSockets, are changing the way businesses interact with customers. Rich communications experiences are being created than enable smart, contextual conversations to take place turning customers into fans. Learn how Digium is engaging with these new technologies as well as how your business can take advantage of web communications.
Phone & Gateway API panel 9:35am - 10:35amDoug Bailey, Digium; Diego Gosmar, Xenialab (xCALLY); Gaston Draque, Nexacomm; Giuseppe Innamorato, Xenialab (xCALLY)This presentation hosted by Doug Bailey, Software Engineer with Digium, introduces the JSON based API of the Digium Gateway and Phones, specifically how the API is structured and what capabilities are available. An example of the API will be shown using sample scripts and the panel guests will discuss the methodology and implementation of the APIs into their solutions.
Xenialab will discuss methodology, integration and configuration of Digium Gateways and phones within their call center solution xCALLY and xCALLY-Shuttle.
Nexacomm will discuss the integration with call-center solutions with very specific needs of building custom apps, mainly into non-DPMA systems. Similar approaches could apply to non call-center scenarios. In addition the main aspects and decisions essential for better integration of these solutions in order to scale properly when middleware is required.
Building A Marketing Machine with UC 11:00am - 11:30amBrian Ferguson, Swvx. Product Marketing Manager, DigiumHaving a strong, reputable brand is vital to keeping and growing your customer database. That’s why large enterprises spend millions of dollars each year to promote and protect their brand. As with most small and medium businesses, you typically don’t have millions to spend – but there are other options. For SMBs, a Unified Communications (UC) phone system is often an overlooked – and affordable – tool to help not only build your brand, but compete with the big guys.
PRE-CONFERENCE DAY: Wednesday, October 22nd
OVERVIEW:
Breakfast 7:30am - 9:00am Summerlin/Charleston Hallway
dCAP Testing (by appt. only)
8:30am - 11:30am Willows
Pre-Conference Sessions
Asterisk from Scratch:Summerlin F 9:00am - 5:00pm Digium Commercial Sessions:Charleston A 9:00am - 4:00pm
Distro Sessions:Veranda C 11:00am - 2:55pm
HackathonSummerlin A9:00am - 5:00pm
Lunch 12:15pm - 1:45pm Pool Area
EXPO Opening Reception 5:00pm - 7:00pm Exhibit Hall - Summerlin B
Fastest Dude Contest 5:30pm - 7:00pm Exhibit Hall - Summerlin B
Hackathon Judging 5:00pm - 7:00pm Summerlin A
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In this presentation join Brian Ferguson, Product Marketing Manager for Switchvox, as he shows you how to use features from an advanced phone system, to build and maintain your brand. You’ll discover how Switchvox can help you:
• Make great first impressions with each incoming call• Tell a more persuasive company story• Get more from your marketing campaigns• Deliver a consistent brand experience from
desktop to mobile• Improve customer service and build better
customer relationships
UC Challenges and Solutions in Education11:30pm - 12:05pmKent Brooks, Director of IT from Casper CollegeTroy Eggleston, Coordinator of Telecommunications Services for Clark County School District Educational institutions face challenging and unique problems everyday, and there is no tool more important for success than a high quality, flexible communications system. A modern Unified Communications (UC) system can create safer, more efficient, more organized, and more productive classrooms, lunchrooms, dorms, and libraries with robust feature sets that benefit every role on campus.
Join us for a conversation Kent Brooks, Director of IT from Casper College and Troy Eggleston, Coordinator of Telecommunications Services for Clark County School District to learn about real life challenges and solutions using Digium products.
Digium Cloud Services 1:45pm - 2:20pmMark Amick, Product Strategy, DigiumStarting with Asterisk, Digium has been at the forefront of disruptive communications technology for more than a decade. Cloud services have been the most recent disruptive force and Digium has become a leader in the industry. Starting with Switchvox Cloud, Digium has been developing Cloud products that are changing the way companies communicate. Join Mark Amick, Director of Product Management at Digium as he tells the Digium Cloud Services story and shares a new product announcement and the future of Digium’s Cloud offerings.
Switchvox Anywhere - Deployment Options 2:20pm - 2:55pmAdam KramerYou can easily future-proof your business for tomorrow. As your business grows, with Switchvox Anywhere, you can seamlessly move to a larger system or the cloud with minimal associated costs and network changes.
Partnering With Digium 3:00pm - 4:00pmJim Butler, Dir. Global Channel Sales, DigiumAs a Digium Authorized Channel Partner, you can sell the full line of Digium’s award-winning Switchvox Unified Communications systems, or a complete solution built on Asterisk, the world’s most popular open source communications engine. Digum has the field-proven solutions you need to win more business.
NOTICE: All AstriCon 2014 conference sessions are being recorded
(video/audio) and broadcast . If you do not wish to be recorded, please sit behind or to the sides of cameras and
refrain from asking questions during sessions .
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Distro Sessions - Veranda C
Using the FreePBX EndPoint Manager 11:00am - 11:30pmLuke Duquaine, Lead Developer, Schmooze Com, Inc.The FreePBX EndPoint Manager can auto provision and configure directly from the GUI interface over 220 popular devices, wireless phones, door phones, paging devices, gateways and specialty devices. Luke will provide an in depth view of managing devices including using Basefile Editor to change settings not exposed in the GUI to fully customize your config files.
Elastix Call Center Pro 11:30am - 12:05pmHideki Harada, Product Manager, ElastixOverview, advantages over the traditional version, and what’s in the road for the new Elastix Call Center PRO.
FreePBX Phone Apps for Digium Phones Presentation 1:45pm - 2:20pmTony Lewis, Schmooze Com, Inc.FreePBX Phone Apps are a suite of IP phone applications that integrate directly with FreePBX and are compatible with devices from several popular manufacturers. These onscreen applications allow users to control functions and settings directly from their phones using apps such as Voicemail, Call Parking, FollowMe Management, Do Not Disturb, Conference Room Management, Login/logout hot desking. This session, facilitated by Tony Lewis, will feature a live demo of several FreePBX Phone Apps using Digium IP Phones, as well as highlight some of the upcoming new apps under development.
Elastix 3.0 2:20pm - 2:55pmElvita Crespo, Channel Manager, Elastix
Key features in the newly released version of Elastix, and their impact in the dynamic VoIP market.
Asterisk from Scratch 9:00am - 4:55pm | Summerlin FMelissa Shepherd: Asterisk Training Manager, DigiumJustin Hester: Asterisk Technical Trainer, Digium
Asterisk from Scratch is the 2014 edition of the wildly popular Asterisk 1-2-3 Seminar. It is a well-rounded and informative overview of the Asterisk Project, with a focus on the essentials a general “newbie” should know. All are welcome! Whether you are managing a team deploying an Asterisk implementation, evaluating potential replacements for a legacy telephony solution, or are simply new to Asterisk, this talk has something to offer you.
This all-day seminar is broken into multiple sessions, beginning with background information such as the history of Asterisk, followed by methods of installing Asterisk and examples of fundamental configuration for call flow and features such as Voicemail and IVR. Lecture will immediately be followed by a Question and Answer Session.
Session 1 (9:00am - 10:30am) Intro to Asterisk & Asterisk Architecture
Session 2 (11:00am - 12:15pm) Installing Asterisk & Intro to Dialplan
Session 3 (1:45pm - 3:00pm) Configuring Endpoints & Interactive Dialplan
Session 4 (3:30pm - 4:55pm) Advanced Dialplan
Hackathon Judging - Wed. Oct. 22nd, 5:00 - 7:00pm, Summerlin AEach team will have 3-5 minutes to give a demo of their app for our panel of expert judges. The top 3 teams will
have the oppertunity to present their demo to the AstriCon audience on Thursday morning. Voting takes place all
day on Thursday. See the bottom of page 17 for more information on the voting process.
KEYNOTE DAY 1: 9:00am - 9:45am | Charleston B
Craig Walker Founder & CEO, Uberconference
Craig Walker has been the leading the revolution of voice and video communications solutions for a number of years, and has an amazing pedigree in founding Dialpad, Yahoo! Voice, GrandCentral Communications and UberConference. Along the way he has
worked with Google on Google Voice, Google Talk, Google Talk Video (does anything scream web communications more than the words ‘google’, ‘voice’, ‘video’?) - so those wishing to get the inside track will be on the edge of their seats as Craig reflects on his involvement in the journey of communications to the web, and shares observations about the way things are heading.
ABOUTCraig Walker brings a broad background as an experienced entrepreneur, venture capitalist, executive and corporate attorney to UberConference, launched in 2012 with an initial $3 million investment from Google Ventures. The idea for UberConference was conceived during the time Craig was Entrepreneur-in-Residence there. After successfully launching the product, Craig raised another $15 million in Series B financing for UberConference from Andreessen Horowitz and Google Ventures.
Prior to UberConference, Craig was CEO and co-founder of GrandCentral Communications. The company was acquired by Google in July 2007, and GrandCentral became Google Voice. While there, Craig ran Google’s Real Time Communications Group (Google Voice, Google Talk and Google Talk Video).
Craig’s career in telecommunications began in 2001 when he ran Dialpad Communications, which he sold to Yahoo! in 2005. He was also a technology venture investor at Sterling Payot Capital and TeleSoft Partners and a corporate attorney at top-tier law firms in Silicon Valley, where he represented companies ranging from early stage startups to Cisco Systems.
Craig is a graduate of UC Berkeley (BA), Georgetown University (MBA) and Boalt Hall School of Law (JD). He is a recipient of the 2014 Georgetown Entrepreneurial Excellence Award.
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CONFERENCE DAY 1: Thursday, October 23rd
OVERVIEW DAY 1:
Breakfast 7:30am - 9:00am Summerlin/Charleston Hallway
dCAP Testing (by appt. only)
8:30am - 11:30am Willows
Keynote Address 9:00am - 9:45am Charleston B
Hackathon Public Voting All-day (see page 17 for details)
Conference Sessions 10:00am - 4:05pm
EXPO Open 10:35am - 5:00pm Summerlin B
Lunch 12:15pm - 1:45pm Pool Area
Fastest Dude Contest 3:30pm - 4:45pm Exhibit Hall - Summerlin B
All-Conference Party 6:00pm - 8:00pm Cherry Lounge (must show your AstriCon Conference Badge to attend)
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CONFERENCE SESSIONS DAY 1: THURSDAY, OCTOBER 23rdDeveloper & Tutorials | 10:00am - 4:05pm | Summerlin A
10:00am - 10:35amTesting with SIPPClod PatryThis session will introduce how to use SIPP. Many integrators or developers are troubleshooting their SIP problems on their network and this software is a perfect tool to replicate some call flows.The session will explain the vocabulary, the exchange of SIP messages and how to create different scenarios for your different needs when troubleshooting your SIP Networks. Clod will also cover a few other useful options when running SIPP that will make your troubleshooting easier.
11:00am - 11:35amBuilding Rich Communications Applications Using ARIMatt Jordan and Samual Fortier-GalaneauJoin Matt and Sam from the Asterisk Dev team at Digium as they explain the depth at which the ARI allows you to connect with Asterisk, and share tips and tricks to make your applications more effective. The ARI is the best interface for a whole range of different applications, and you’ll want to attend this session to see for yourself how much more comprehensive the ARI is than any of it’s predecessors, and to get a steer from the experts on how to go about building your applications.
11:40am - 12:15pmScaling FastAGI Applications with GoLefteris ZafirisA presentation about combining the powers of the Go language and Asterisk in order to provide fast, reliable and hugely scalable voice applications. A brief introduction about why Go presents a big opportunity for the asterisk community, a primer into developing FastAGI applications, demonstration of the AGI package [1] and a walk through the idioms and challenges of developing in Go. The talk is focused mainly on our experience of porting existing code into Go with the aim to scale our services to larger numbers accompanied with benchmarks and an introduction to some tools [2] we developed to help us test, debug and benchmark AGI applications.
1:45pm - 2:20pmTwelve Reasons Your API SucksKeith CaseyThose first moments of using an API are pivotal. There’s nothing like downloading this week’s PDF of the documentation, setting up a SOAP client, reconfiguring all your URLs, and decoding the latest binary payloads. It makes your heart sing and your blood pressure rise. Just like there are code smells through the rest of your project, there are API smells that make them hard to design, hard to launch, and hard to maintain. We’ll use this time to explore a few common APIs to highlight those issues and demonstrate strategies to fix the issues before they become problems.
2:25pm - 3:00pmDistribution, redundancy and high availability using OpenSIPS Ali PeySession will cover Redundancy, Load balancing, Distribution and High availability for Hosted, Enterprise and Cloud solutions with multiple telephony gateways such as Asterisk, interfacing multiple carriers, SIP trunks and various SIP Clients such as SIP Phone, Mobile Apps and WebRTC.
3:30pm - 4:05pmGetting the Best out of WebRTCTim Panton and Dan JenkinsThis talk is aimed at giving you the tools to help you scale your web webRTC deployments while still meeting your user’s needs.Session will cover:• User expectation • Topology • Monitoring• Selecting your codec • Tuning your codec • Auxiliary servers STUN/TURN/Proxy • Crypto choices • Transcoding
We will attempt to answer these questions and provide the audience with resources to help them scale their webRTC deployments. Where possible we will use graphs and demos to illustrate the trade offs.
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CONFERENCE SESSIONS DAY 1: THURSDAY, OCTOBER 23rdBusiness & Case Studies | 10:00am - 4:05pm | Charleston A
10:00am - 10:35amBuilding a Conference Bridge System on AsteriskMartin KaplanThe intention of this presentation is to show to the au-dience, that Asterisk can succeed in companies where there is already an existing commercial PBX system. The session will cover creating a conference bridge on Asterisk with basic web admin tool using PHP/AMI/AGI and making a SIP trunk with Virtual Avaya PBX with a connected Avaya phone.
11:00am - 11:35amMaintaining an International Multi-Site Asterisk NetworkCassius SmithIn 2010 TransWorld Radio (TWR), a global nonprofit media ministry, began replacing its mélange of pro-prietary PBX systems with Asterisk. Since then the pri-mary challenge has been maintaining and updating/upgrading the distributed systems over 16 time zones. This session is of interest to anyone maintaining a multi-site deployment of Asterisk, especially on a lim-ited budget, and how to choose the distribution (build from source, FreePBX, Elastix) for ease of maintenance.
11:40am - 12:15pmAsterisk at the Heart of Interactive MediaSebastien CoutureDigital media is in the midst of a revolution. As social media, smart phones, and smart TVs become more ubiquitous, they also allow for countless new and in-ventive ways for consumers to experience and engage with content such as TV shows, advertisement or art.
As an open source toolkit, Asterisk is the perfect en-gine to an interactive media framework where tech-nologies such as telephony, text messaging and voice recognition are needed.
This presentation will showcase “Emilie”, a Canadian interactive web series that went on to win several awards, including one for Best TV Series, Web & Trans-media at the Geneva International Film Festival, as well as a nomination at the SXSW Interactive Awards.
1:45pm - 2:20pmMarketing Boot CampSusan ElderYou know you need to market your business, or even yourself, but are you unsure where to start? Do you think that marketing is expensive and requires a big budget, which you don’t have, so you don’t do any marketing? Are you not sure where to start and how to prioritize marketing activities? This Marketing Bootcamp session will walk you through how to draft an action-oriented marketing plan emphasizing free or low-cost resources, including manufacturer and dis-tributor marketing portals, electronic marketing, social media, public relations and more. As part of this ses-sion, attendees will receive a marketing plan template.
2:25pm - 3:00pmThe New Rules for IVRAllison SmithForget everything you’ve been told about painstaking-ly drawing up a complicated IVR schematic; ditch the golden rule about how many options you need to pro-vide, and please – stop telling the caller how valuable their time is while they’re wasting it on hold. There are new standards for designing your call flow; there are fresh ways of viewing your IVR and its potential; and there are ways to enhance your caller’s experience as they move through your phone system. Allison Smith – The Voice of Asterisk – through her years of experi-ence voicing prompts for Asterisk and multiple other systems – can show you the common pitfalls of IVR design and how to avoid them -- and how to use “The New Rules for IVR” to make your system as intuitive and friendly as it can possibly be.
3:30pm - 4:05pmTelecomunicaciones Abiertas de Mexico: How we increased Safety for Journalists & Human Rights DefendersJuan Carlos FernandezDue to the increasing insecurity faced by journalists and human rights workers in Mexico, the Mexican Government passed a law in 2012 calling for the rapid implementation of measures to help protect them. In response, Telecomunicaciones Abiertas de Mexico provided an innovative, practical, reliable, and cost-effective solution using an Asterisk-based system in conjunction with other technologies. The system was in place and operational by the end of 2012 and has been used successfully to defuse dangerous situa-tions to its users.
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CONFERENCE SESSIONS DAY 1: THURSDAY, OCTOBER 23rdSecurity and Asterisk-based Distributions | 10:00am - 4:05pm | Veranda DE
10:00am - 10:35amMaking your Asterisk System SecureEric KleinOver the past few years Eric has shown that telecom fraud is a growing problem, and basic fixes for protect-ing your (and your customers) PBX. This time he will show the basic configuration considerations that you can take to protect a PBX. Come to this session to find out: Who is out there looking to attack your PBX? How do they find it? How can you protect your PBX?
11:00am - 11:35amUpdates in Fraud Protection and PreventionFlavio E. GoncalvesThis presentation covers Asterisk security and fraud prevention. The objective is update the audience in tools and techniques used in fraud attempts. We will start with a short review of the last year’s presentation and show the newest trends observed from fraudsters. Most of the information was gathered from our honey-pots. The proposed presentation agenda is:
• Shost review of last year’s presentation• Country distribution of the attacks in 2014• Tools distribution of the attacks in 2014• Main Targets.• New tools being used• New methods• New target URLs• Impact of the heartbleed vulnerability• IAX2, SKYNY, H323, TFTP possible vulnerabilities• How to detect and prevent
11:40am - 12:15pmVoice & Security: What’s Happening in the Real WorldErik KnightThere are number of things people are hearing about in VoIP security but not not sharing: from toll fraud, securing your clients network and the liabilities as-sociated with having a phone on someone else’s network. Do you trust the phone manufactures? Are you liable for a premise based info-structure. We’ll dig down into many of these things including securing the voice stream and good practices. We would really like to interact with the crowed and speak to their talking points.
1:45pm - 2:20pmVoIP Security: Common Mistakes, Prevention Methods and Unforeseen Attack VectorsPhillip MullisYour invited to see the many common mistakes and often overlooked areas of security in locking down your installations and infrastructure. Witness items and system hacks of absolute nightmare and panic. Many areas are touched upon from dial plan to web exploits and some extra fun areas that many have no clue are being hit!
2:25pm - 3:00pmSecurity PanelChris Brown, Flavio E. Goncalves, Eric Klein, Phillip MullisCome along with your toughest security questions! Our panel of Asterisk Community members, all of whom have solid knowledge and experience in the area of Security, will be ready to answer!
3:30pm - 4:05pmFreePBX Yesterday, Today and TomorrowPhilippe LindheimerFind out how FreePBX® grew to be more than just an easy way to write config files for Asterisk®, to beco-methe most widely adopted Open Source PBX Platform and EcoSystem in the world, we will provide an over-view of what the FreePBX® project is and a timeline in history of how FreePBX has evolved into what it is today, including how our millions of users benefit as well as a sneak peak at what is coming next.
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CONFERENCE SESSIONS DAY 1: THURSDAY, OCTOBER 23rdService Provider and Contact Center | 10:00am - 4:05pm | Summerlin F
10:00am - 10:35amSystem Monitoring Enterprise Asterisk ClustersJames PearsonThis session will provide an overview of some of the tools available to monitor a large Asterisk cluster with hundreds of servers using available open source tools and Asterisk itself to help in the notification and monitoring of system resources, networks and carriers.
11:00am - 11:35amGetting Started with WebRTCBilly ChiaWebRTC “Web Real-Time Communications” is an innovative technology that is poised to change the communications landscape. In this talk Billy will explain WebRTC, highlight some of the trends and use cases in web communications and demo some WebRTC video calling functionality.
11:40am - 12:15pmContact Center Scaling with AsteriskCorey McFaddenWhat do you do when you need to support hundreds of contact center agents? Not only can Asterisk can help--it can do so cost-effectively. (This session is for both business and technical users.)
Explore concepts including:• Multi-node architecture• Virtualization• Queue and agent state federation• Redundancy & High-Availability• Reporting & Analytics
1:45pm - 2:20pmFronting Asterisk with Kamailio for WebRTC and web-service integrationPeter DunkleyTwo important aspects for providing any service are scaling and security. This session will explain how Kamailio can be used to distribute traffic across many Asterisk instances (for scaling), how to configure Kamailio to receive SIP over WebSocket traffic (for WebRTC), and how to authenticate this traffic in a way that integrates with a web-service (for security).
2:25pm - 3:00pmAutomatic Configuration Management for Kamailio and Asterisk in the era of PuppetGiacomo VaccaWe are under the pressure of delivering RTC systems that are at the same time stable, but can change often to add features and fix bugs. The underlying systems also change frequently (OS upgrades, increase/decrease capacity on demand), and we need multiple environments running (e.g. development, testing/QA, production). Puppet provides a solution that dramatically cuts deployment time, reduces occurrences of errors, while at the same time documenting the configuration status.
This presentation is about how I stopped worrying about the deployment of systems built with Kamailio, Asterisk and other open source applications. With our Puppet code base we go from a new Virtual Machine to a working system in minutes, with piece of mind and self-documented configurations and processes. Firewall, nagios, syslog, monit, sec, and many other related services and properties are also automatically configured.
3:30pm - 4:05pmKamailio - SIP Servers EverywhereDaniel Constantin-MierlaKamailio, the open source SIP server, has just entered the 14th year of development. Along with the update on latest additions, the presentation is going to focus on the benefits of deploying a pure SIP signaling server and what are the typical as well as unexpected and surprising use cases, from routing engine for large, scalable IP Telephony or WebRTC services, to dedicated roles on embedded or tiny IoT devices.
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Our Greatest Creation Yet!
Visit us at booth 27 to discover what’s new in FreePBX 12,and schmoozecom.com to learn how to get your piece of the Billion Dollar Open Source Telephony Marketplace.
FreePBX 12, the most significant leap forward with huge internal upgrades that paints the canvas for improved
functionality and new features for years to come.
All-day Thursday: Hackathon Voting - Vote for your favorite!On Thursday everyone is invited to vote for the Popular Choice winner in the AstriCon Hackathon.
Go to http://bit.ly/AstriConVote and cast your vote for your favorite hackathon app!
All of the submissions can be seen and voted on from the AstriCon Hackathon Challenge Post page: http://astriconhackathon.challengepost.com/submissions
Instructions for voting here: http://help.challengepost.com/How-to-vote-for-a-submission.html
Voting is open world-wide, so share and solicit votes for your favorite hack on Twitter, Facebook, and LinkedIN!
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CONFERENCE DAY 2: Friday, October 24th
OVERVIEW:
Breakfast 7:30am - 9:00am Summerlin/Charleston Hallway
dCAP Testing (by appt. only)
8:30am - 11:30am Willows
Keynote Address 9:00am - 9:45am Charleston B
Conference Sessions 10:00am - 3:00pm
EXPO Open 10:35am - 2:00pm Summerlin B
Lunch 12:15pm - 1:45pm Pool Area
Asterisk Project Updates/ AstriCon Recap 3:00pm - 3:30pm Charleston B
Wrap Party BBQ 4:00pm - 6:00pm Cherry Lounge (must show your AstriCon Conference Badge to attend)
KEYNOTE DAY 2: 9:00am - 9:45am | Charleston B
Russell BryantPrincipal Software EngineerRED HAT
Can you imagine someone better placed to talk about how communications can be best closeted in the cloud and wired to the web than a guy that had a leadership role in the Asterisk project before moving to Red Hat to work on OpenStack and becoming a an elected member of the
OpenStack Technical Committee? That guy is Russell Bryant and we are very pleased to say that Russell is still involved in Asterisk and has served as the OpenStack Compute (Nova) Project Technical Lead for two releases (Havana and Icehouse), and he is still a member of the Nova Core team. This places him very well indeed to share his views on where we are and his perspective on opportunity for further innovation.
ABOUTRussell is a Principal Software Engineer at Red Hat where he works on the OpenStack project. He is an elected member of the OpenStack Technical Committee which provides overall technical oversight and governance for the OpenStack project. Russell served as the OpenStack Compute (Nova) Project Technical Lead for two releases (Havana and Icehouse) and is still a member of the Nova Core team.
Russell also has a long history in the Asterisk community. He has been contributing to Asterisk development in some capacity since 2004, including leading the project for some time. Russell co-authored “Asterisk: The Definitive Guide” published by O’Reilly Media. He also contributed a chapter on Asterisk to the book “Architecture of Open Source Applications”.
NOTICE: All AstriCon 2014 conference sessions are being recorded
(video/audio) and broadcast . If you do not wish to be recorded, please sit behind or to the sides of cameras and
refrain from asking questions during sessions .
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10:00am - 10:35amWebRTC: The Big Debate, Shut Up and Build SomethingLantre BarrWeb Real-Time Communications (WebRTC) has been rapidly growing for the past 3 years gaining adoption from many corporations and industries around the world. Surpassing the one billion mark for compatible devices, WebRTC has sparked some heated debate around whether or not it’s a truly disruptive technol-ogy. In this presentation we will discuss the possibili-ties and the current applications built and running full WebRTC services, despite the lack of support in Internet Explorer and Safari and the codecs wars.
11:00am - 11:35amIntroducing Payload-Voice: Call Queues for the Asterisk REST InterfacePaul BelangerPayload Voice is the result of last years Astricon talk “Distributing call queues for Asterisk using Python and Redis”. Since then we’ve forged a head writing code to replace app_queue in Asterisk. In this talk we’ll ex-plain how the application work and demo call queues with ARI.
11:40am - 12:15pmConnecting Non-SIP IP Camera to Your PBXTom De MoorConnecting a SIP-enabled ip camera to Asterisk is easy. But do you know that you can also connect the rtps to your Asterisk?
In this session you’ll see some techniques to connect those camera’s to your PBX. You’ll be able to make a videocall to your camera and move the camera around with your keypad. Combine this with WebRTC and 4G-connections and you get new possibilities in securing buildings. 1:45pm - 3:00pmDangerous DemosJames BodyDangerous Demos: We would like to run an up-gunned version of DD similar to that run at Kamailio World this year. Format would be a minimum of a 1 hr slot, with a minimum of 8 contenders, each present-ing their DD in a slot taking no more than 4 minutes (allowing circa 90 secs for each change over). Win-ners determined by live audience voting (powered by Asterisk). Prize(s) supplied by Truphone.
CONFERENCE SESSIONS DAY 2: FRIDAY, OCTOBER 24thDeveloper & Tutorials | 10:00am - 3:00pm | Summerlin A
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10:00am - 10:35amFreePBX High AvailabilityTony LewisFreePBX HA is a High Availability solution that enables automatic mirroring and failover between two FreePBX systems. It was created to fill a need for organizations that have a low tolerance for downtime in the event of system failures and outages. Tony will review how it works, and provide a live demonstration of FreePBX HA in action in conjunction with a Digium R Series Appliance. FreePBX HA recently was awarded the Best Large Enterprise Application at ITEXPO Miami.
11:00am - 11:35amCloud Based Asterisk - Tips from the Front LineNir SimonovichAmazon, Google Compute Engine, Rack Space, Digital-Ocean - wow, so much to choose from, but yet, very little information is known as to how to bring your Asterisk installation to the max in these environments. Let’s explain some basics about some of these, and explore the differences and the pros/cons for each.
11:40am - 12:15pmAsterisk Collaboration on Web-Functions and Adapt to Carrier - Grade SpecificationsTakahiko NagataWe made asterisk as a platform which is able to col-laborate with Web Services on the internet by enhanc-ing asterisk in the following aspects: - expose REST APIs - connect with API management productsBy using our platform, you can provide video contact centre which can handle WebRTC calls in addition to traditional SIP/PSTN calls. We will go through how we could apply asterisk on carrier network.
1:45pm - 2:20pmHow to Sell it Better: A Case of WebRTC in RetailJakub KlausaThis session will show both the technical (mainly) and business aspect of the ongoing project to implement WebRTC based ‘sales concierge’ for an electronic retail chain.
2:25pm - 3:00pmCreating ARI apps FAST!Matt RiddellExamples on how to create applications that talk to Asterisk via the Asterisk Rest Interface. A simple way to get started developing applications that work with Asterisk. Primarily focussing on PHP using the PHPARI library.
CONFERENCE SESSIONS DAY 2: FRIDAY, OCTOBER 24thBusiness & Case Studies | 10:00am - 3:00pm | Charleston A
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10:00am - 10:35amWhat’s In Asterisk 13?Matt JordanAsterisk 12 was really a step change from previous versions in terms of internal changes made to Asterisk to enable a clear path for progression in the future. Several key pieces of deep functionality were totally re-engineered and a brand new SIP stack was added. Asterisk 13 (an LTS) represents the maturing of the innovative things that were introduced in Asterisk 12, and this session will give you low-down on all the new goodies that are now in a production-ready state. Don’t miss this one!
11:00am - 11:35am¡Viva la Revolución!Jim “Dude” DixonA brief view of the history of the revolutionary changes in the telecom industry and technology, and how Asterisk/Zapata Telephony made it happen.
11:40am - 12:15pmUsing Asterisk to create “Her”Ben Klang and Allison SmithIn the film “Her” the protagonist falls in love with his computer, an artificial intelligence operating system. While most of us already love Asterisk, things really get interesting when we give Asterisk a voice, and the ability to listen to our instructions.Fortunately for us, Asterisk has impressive capabilities for adding speech recognition and text-to-speech to our calls. This talk will cover many facets of speech applications with Asterisk. We will look at the various commercial and open source speech engines available, as well as how to integrate them into Asterisk. We will look at ways prompts and grammars can be designed to give the caller the best possible experience. We will hear samples of the right and oh-so-very-wrong ways speech can be used. We will cover the various types of speech recognition that exist today (grammar-driven, transcription, hotword and voice biometrics) and how each should be applied.
Finally, we’ll show how these pieces come together to make it possible to build something that (for a brief moment) passes as intelligent. Maybe.
1:45pm - 2:20pmOpenSIPs for Asterisk UsersPeter KellyA gentle introduction to SIP Proxies from an Asterisk users perspective. The idea is to give Asterisk users who have a single server install some ideas about how they could improve resilience and take steps to increase capacity should they need to.
2:25pm - 3:00pmAsterisk, Elastix,and FreePBX ClinicScott Griepentrog and Mark Michelson, AsteriskElvita Crespo and Jorge Gutierrez, ElastixTony Lewis and Andrew Nagy, FreePBXWe bring you face to face with the people behind the projects! If you have a query about Asterisk, Elastix or FreePBX, then attend this clinic, where members of the development team from each project will be on hand to assist.
CONFERENCE SESSIONS DAY 2: FRIDAY, OCTOBER 24thSecurity and Asterisk-based Distributions | 10:00am - 3:00pm | Veranda DE
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CONFERENCE SESSIONS DAY 2: FRIDAY, OCTOBER 24thService Provider and Contact Center | 10:00am - 3:00pm | Summerlin F
10:00am - 10:35amTipping the ScalesBen KlangThis is where we attempt to answer the oft-heard question: “Can it scale?”
Floating around the Internet are plenty of benchmarks and wildly varying of numbers that claim Asterisk can support some number of calls. But these numbers are often hiding important assumptions, and they rarely serve as a good predictor for the performance of a given deployment. What then should you measure, and how will you know what kind of load a given server can handle? What is the difference between concurrency and call rate, and when should you consider each? The real numbers come from how the telephony engine will actually be used. This talk will look at the various ways Asterisk gets used in the real-world, and how various features (passthrough, transcoding, recording, playback, and conferencing) impact the capacity of a given server. We will also look at some of the tools available for generating load and measuring the performance.
11:00am - 11:35amInterfacing with Asterized Internet of Things NodesMichael Schloh von BennewitzThe Internet of Things is creeping into legacy systems and in this hour we examine what role telephony service and client applications can play in the IoT. A quick tour of definitions, market movements, and industrial strategy sets the stage for deeper inspection of IoT technology. We log into a low power embedded prototyping board used in IoT research (a grant of either Galileo or Edison from Intel) and snoop around the file system, finally deploying Asterisk and quickly tuning for the unique IoT platform. An in depth demonstration of a home made Asterisk module to
drive the Arduino-compatible shield assembly of a Galileo prototyping board further illustrates theory and concepts of the first half hour. Code samples of AMI and MQTT based communications between IoT nodes are explained and related to SIP signalling. Finally, a strangely relevant 2-3 minute Hollywood film clip adds a fun ending to the presentation while preparing the mood for questions.
11:40am - 12:15pmIntegrating Asterisk with a feature-rich browser UIPaul AshfieldThe secret to driving adoption and ensuring platform usage is to provide features that are accessible, easy and fun to use. In this hands-on session we will build a browser-based, responsive UI to view and control Asterisk conferences and calls. We’ll also demonstrate how this approach can be extended to cover provisioning and management of IPT services.
1:45pm - 2:20pmDeveloping an IVR Payment System with AsteriskKayode AkintundePresentation will be based on an IVR payment architecture i just completed for one of the leading Payment Switching Company in Nigeria.
2:25pm - 3:00pmService provision using Asterisk and OpenSIPSBogdan-Andrei IancuOpenSIPS and Asterisk are a great combination to en-able all sorts of SIP services. Starting from architectur-al level (like scalability, redundancy, geo-distribution), all the way up to advanced services (like LCR, SIMPLE Presence, Call Centers), OpenISPS is the Asterisk com-panion. The typical OpenSIPS sandwich with Asterisk is the perfect combination of class 4 and class 5 SIP tools in order to achieve the ultimate platform.
SPECIAL BREAKOUT SESSION Asterisk Project Updates and AstriCon Recap
3:00pm - 3:30pm, Charleston B
Immediately following this session is a Wrap Party BBQ in the Cherry Lounge.
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Ali PeySenior Software Engineer, OpenSips
Ali Pey is a Sr. Software Engineer Architect with more than 22 years experience in Telephony, Networking and VoIP. He has an Electronics Engi-neering degree with a focus in tele-
communication and software design. He has worked for companies such as Nortel and TalkSwitch and has been developing VoIP solutions since the start of the technology. He has developed software for Proxy Servers, Registrar Servers/Clients, User Agents, and other VoIP components in both SIP and H.323 proto-cols. Ali currently works for j2 Global (j2.com), a global company for Cloud Services and has successfully used Asterisk, OpenSIPS and other open source platforms to provide Global Telephony Cloud solutions for j2.
Allison SmithVoice Goddess, The IVR Voice.com
Allison Smith – The Voice of Asterisk – is one of the most prevalent tele-phone voices working today, having voiced platforms for major telcos as well as companies as diverse as EBay,
Victoria’s Secret, Halliburton, 3M, and PetSmart, to name a few. As well as being automatically integrated into your Asterisk system via the stock prompts, Al-lison records custom prompts for the Asterisk Com-munity every day; for industries as varied as public utilities, justice system auto-dialers, large retail chains, non-profits, military, and pharmaceutical. Known for her lightning-fast turnaround, her versatility, and her passion for voicing IVR, Allison is especially fond of the Asterisk community, and looks forward to serving it for years to come.
Ben KlangFounder, Mojo Lingo
Ben Klang has been involved in the Open Source community since 1996, when he first installed Linux for his high school’s web server. Since then, he has contributed directly to many
open source projects, including Asterisk, Horde, ldap-2dns, and many others and most recently has taken over leadership of the Adhearsion project. With expe-rience in diverse technology disciplines such as tele-phony, system administration, software development and networking, he enjoys solving broad problems. He founded a company, Mojo Lingo, in 2007 with a mis-sion to create voice applications that work like magic. His interests include hacking competitions, informa-
tion security, making phones do unexpected things, playing music loudly and asking lots of questions.
Brian FergusonProduct Manager, Switchvox, Digium
Brian Ferguson is the Product Mar-keting Manager for Switchvox. Since becoming part of the Digium team in 2010, Brian has worked in multiple sales and marketing roles supporting
Diguim’s reseller channel and helping SMB customers use Digium solutions to solve their business tele-phony challenges. With an additional 10 years of sales management experience in a retail environment, Brian has extensive experience working with a diverse range of technical products and brings a unique perspective in helping problem solve for customers. He graduated with honors with a BS in Business Management from the University of Alabama at Huntsville.
Cassius SmithSenior Project Consultant, TWR (Trans World Radio)
Cassius Smith has been with TWR since June 2010 as a second career missionary along with his wife, Valerie. Cassius retired in 2009 after 32 years
as a software developer and project manager in the aerospace industry. Cassius is a Senior Project Con-sultant at TWR and works in the Cary, NC office. As the resident dCAP, he is also responsible for deployment and maintenance of the worldwide network of Asterisk systems, which support TWR’s regional offices in the Americas, Europe, Africa and Asia.
Chris BrownSenior Director of Information, Vitelity and Onvoy Voice Services
Christopher Brown, has provided consulting services for several Fortune 500 companies, major telecommuni-cations providers, and law enforce-
ment organizations for the past ten years, focused on hacking activities, phone phreaking, and system design. He is an active member of the Communica-tions Fraud Control Association (CFCA) and often speaks at their meetings and events. Christopher was the lead engineer on the Open Source project trixbox, a softswitch based around Asterisk PBX. Christopher has worked for several ISTP’s, and most recently devel-oped the voice platform for Vitelity’s award-winning vMobile product. Christopher is currently running Product Research, Professional Services, and the Fraud
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Department for Vitelity and Onvoy Voice Services where his acting title is Senior Director of Information.
Clod PatryAsterisk Enthusiast, AMUG
Clod Patry has worked full-time with Asterisk for over 7 years. He has many contributions to the code and is currently working with SDP.
Corey McFaddenFounder & President, Voneto LLC
Corey McFadden is the founder and President of Voneto LLC, a commu-nication solutions firm that ranks among the top national IP telephony integrators. He brings over 18 years
of industry experience to Voneto with a unique back-ground that includes experience in call center opera-tions, field-consulting, project management, IP tele-phony, and software development. Corey has been a featured lecturer at numerous business conferences, published articles on business and technology, and been the subject of a television program profiling visionary businesspeople in Pennsylvania.
Craig WalkerCEO & Founder, UberConference
Craig Walker brings a broad back-ground as an experienced entrepre-neur, venture capitalist, executive and corporate attorney to UberConfer-ence, launched in 2012 with an initial
$3 million investment from Google Ventures. The idea for UberConference was conceived during the time Craig was Entrepreneur-in-Residence there. After suc-cessfully launching the product, Craig raised another $15 million in Series B financing for UberConference from Andreessen Horowitz and Google Ventures.
Prior to UberConference, Craig was CEO and co-founder of GrandCentral Communications. The company was acquired by Google in July 2007, and GrandCentral became Google Voice. While there, Craig ran Google’s Real Time Communications Group (Google Voice, Google Talk and Google Talk Video).
Craig’s career in telecommunications began in 2001 when he ran Dialpad Communications, which he sold to Yahoo! in 2005. He was also a technology venture investor at Sterling Payot Capital and TeleSoft Part-ners and a corporate attorney at top-tier law firms in Silicon Valley, where he represented companies rang-ing from early stage startups to Cisco Systems.
Craig is a graduate of UC Berkeley (BA), Georgetown University (MBA) and Boalt Hall School of Law (JD). He is a recipient of the 2014 Georgetown Entrepreneur-ial Excellence Award.
Dan JenkinsSoftware Engineer, Digium
Dan Jenkins is a Software Engineer at Digium, specialising in web technolo-gies and services, creating RESTful APIs and bringing VoIP to the web platform. Dan founded his own web
consultancy firm nimbleape.com
Daniel Constantin-MierlaKamailio, SIP and VoIP Consultancy, Asipto.com
Leading Kamailio (formerly named OpenSER), an Open Source SIP Server targeting large deployments, project co-founded in 2005 as a spin off of
SIP Express Router (aka SER).
Involved in VoIP world starting with 2002 as core developer of SER, continuing with lots of activities in development and management of Open Source VoIP projects, bringing Kamailio up to InfoWorld award Best Of Open Source Networking Software 2009.
Running a consultancy business at Asipto.com, Berlin, Germany, focusing on design and implementation of new IP communication services using SIP and Web technologies as well as trainings and development services for Kamailio SIP Server.
Open source advocate and casual blogger.
Diego GosmarCEO, Xenialab (xCALLY)
Diego Gosmar started to work in the Voice over IP market during the early 1997, leading some research projects for Telco carriers.
They approached the Asterisk development in 2004 and in early 2007, together with a team of passion-ate VoIP experts, they exploit the world’s most widely used open source telephony platform, Asterisk, to launch a flagship software solution for the global Contact (Call) Center market called xCALLY. Founded in the technology department at the Polytechnic University of Turin, the xCALLY team is currently led by Diego Gosmar CEO, and Giuseppe Innamorato, CTO and product manager for Xenialab.”
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Doug BaileySoftware Engineer, Digium
Doug Bailey is a Software Engineer at Digium, Inc. He has worked on a number of embedded systems for Digium including the Gx00 series of T1/E1 gateways. Doug has a 30 year
experience base on working on media products includ-ing 8 years at Digium with Asterisk based systems and 15 years in the video production. He has worked on all aspects of product development from low level as-sembly code to web design implementation. Doug has a masters degree in computer science from University of Alabama in Huntsville and a bachelors degree in engineering from Brown University.
Elvita CrespoTelecom Professional, Elastix
Elvita Crespo is a telecom professional that has been involved with Elastix since 2009. She currently works as Channel Manager for the company, fo-cusing on the expansion of its partners
network around the globe.
Eric KleinSr. Consultant / VP Sales and Marketing, Greenfield Technologies Ltd
Eric has more than 23 years of ex-perience in the Telecom industry. In addition to his experience in VoIP security at Greenfield Tech and fraud
prevention at Humbug Telecom, he has worked for carriers MCI Communications (now part of Verizon) and Cellcom. He has co-authored RFC4864 - Local Network Protection for IPv6 and served as a grant reviewer for the US Department of Commerce Broadband Initiatives (BIP) Program and Broadband Technology Opportuni-ties Program (BTOP).
Erik KnightFounder and CEO, SimpleWan
Erik Knight is the founder and CEO of SimpleWan. Current IT Chair & Board member of the Arizona Science Center. Prior to SimpleWan he started COM-VOICE, an early to market hosted VoIP
company, and also an IT company; both successfully sold and rolled up by other companies. He also devel-oped DirtSearch.org, which is a free background search site. Erik currently has 6 patent pending; all involving development and inventing new technologies. His spe-cializations & experience include: Oracle DB; MYSQL/
PHP; Cisco CCNA; Microsoft; Windows/Servers; and FreeBSD/Centos/Linux/Solaris. Summed up Several Database, Networking and Operating System Certi-fications. With programing experience in over eight computer languages. Erik’s technologies and security experience has been reviewed by and used by fortune 500 companies and government organizations. He is a strong believer in investing back into the community. His philanthropy includes: Operation Home Front, St Joseph Hospital, Toys for Tots, Christmas Angel Pro-gram, ChildHelp USA, Make a Wish Foundation, Team Lizzy Bell, The Cade Purdue Foundation, CRS (Chil-dren’s Rehabilitation Services), and many more.
Flavio E. GoncalvesCEO, SipPulse Routing and Billing Solu-tions for SIP
Flavio E. Goncalves is the author of two books, Building Telephony Sys-tems with OpenSIPS and Configuration Guide for Asterisk PBX. He is also one
of the first Digium Certified Asterisk Professionals, achieved in May 2006. He is currently CEO of SipPulse Routing and Billing Solutions for SIP, a company that specializes in SIP softswitches.
Francesco PriorSoftware Developer, Freelancer
Francesco Prior is Italian, has lived many years in Angola, and now re-sides in Argentina. Through these experiences, he has learned to speak Portuguese, Spanish and English. He
is a self-taught person with 10+ years of program-ming experience and a passion for learning. Francesco discovered Asterisk in 2009 and received his dCAP cer-tification in July 2012. He has spent the last two years developing irVoice [www.irvoice.com.ar], a web appli-cation for call centers. He is now a freelancer eager to work, learn and enjoy as Asterisk consultant.
Gaston DraqueVoIP Development and Deployment Spe-cialist, Nexacomm
Gaston Draque is a VoIP development and deployment specialist at Nexa-comm, an IP Telephony supplier based in NYC.
He has been working with company’s information systems for the last 20 years and with Asterisk since 2005. Combining web, database and software engi-neering best practices with the vast business possibili-ties that Asterisk provides, has been a perfect mix for
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this business oriented professional.
Gaston has a B.S in Computer Information Systems, and a B.S in Business Administration from Universidad de Buenos Aires. He is an Associate Professor of “IT Resourc es Administration” at Universidad de Buenos Aires.
Giacomo VaccaSenior Network Applications Developer, Truphone
Giacomo Vacca holds a Master Degree in Electronic Engineering. He’s been working with VoIP and RTC technolo-gies for more than a decade, even
before SIP became a de-facto standard.
With a passion for RTC technologies, Open Source, Linux and Automation, Giacomo leads the Network Ap-plications team at Truphone Labs
Giuseppe InnamoratoCTO, Xenialab (xCALLY)
Diego Gosmar and Giuseppe Innam-orato started to work in the Voice over IP market during the early 1997, lead-ing some research projects for Telco carriers.
They approached the Asterisk development in 2004 and in early 2007, together with a team of passionate VoIP experts, they exploit the world’s most widely used open source telephony platform, Asterisk, to launch a flagship software solution for the global Contact (Call) Center market called xCALLY. Founded in the technol-ogy department at the Polytechnic University of Turin, the xCALLY team is currently led by Diego Gosmar CEO, and Giuseppe Innamorato, CTO and product manager for Xenialab.”
Hideki HaradaVoIP consultant and product manager,Elastix North America
Hideki Harada is a VoIP consultant and product manager for Elastix North America. He has been working with VoIP distribution and consulting for
the last 10 years, dedicating his passion for emerging technologies and IP solutions. Hideki has a Bachelor degree in Business Administration and was born in Venezuela.
Jakub KlausaCTO, SS7 Technologies
Jakub Klausa has worked in the tele-com industry for over 10 years and is always interested in emerging trends in the technology market. He was on the team to build one of the first all-IP
carrier grade networks in Poland. Jakub has experience with both carrier level technologies and consumer accessed solutions. He is currently working with SS7 Technologies, where he’s responsible for the technical projects that make convergence not only a buzzword, but also a reality.
James BodyHead of R&D, Truphone
James Body is currently employed as Head of Research & Development at Truphone, the Global Mobile Network Operator. He is a ‘Veteran’ telecom-munications guy, whose long career in-
cludes some 22 years as a Regular Officer in the Royal Corps of Signals (UK Army) and more recent involve-ment with VoIP and other real-time communications systems. As an enthusiastic supporter of open source applications, he has been responsible for a string of novel and innovative projects, many of which straddle the boundary between the ‘Old School’ telecommu-nications world and more modern application-based services.
James PearsonDirector of Hosting, ViciDial Group
James Pearson has over 13 years expe-rience developing, implementing, and managing Linux and network envi-ronments for small to medium sized business’. Projects ranged from simple
Linux routers with NAS to resilient network design, integrated Asterisk and CCTV solutions, custom devel-oped applications, and IT services management. James is currently the Director of Hosting for the ViciDial Group and manages a colocated hosting platform of over 300 servers for vicihost.com.
Jim “Dude” DixonConsultant, Zapata Telephony Project
The Zapata Telephony Project was con-ceived of by Jim Dixon, a telecommu-nications consulting engineer who was inspired by the incredible advances in CPU speeds that the computer indus-
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try has now come to take for granted. Dixon’s belief was that far more economical telephony systems could be created if a card existed that had nothing more on it than the basic electronic components required to interface with a telephone circuit. Rather than hav-ing expensive components on the card, digital signal processing (DSP) would be handled in the CPU by soft-ware. While this would impose a tremendous load on the CPU, Dixon was certain that the low cost of CPUs relative to their performance made them far more at-tractive than expensive DSPs, and, more importantly, that this price/performance ratio would continue to improve as CPUs continued to increase in power.
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. 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:
Over the years, the Zapata Telephony interface in Asterisk has been modified and improved. The Digium Asterisk Hardware Device Interface (DAHDI) Telephony interface in use today is the offspring of Jim Dixon’s contribution.
Jim ButlerDirector of Global Channel Sales, Digium
Jim Butler is Director of Global Chan-nel Sales at Digium, Inc. He has 30 years of experience in channel sales and management for voice and data networking companies including US
Robotics, Motorola, ADTRAN, and Digium. In all ca-pacities, Jim has been responsible for designing and implementing best-in-class channel programs which focused on growing top and bottom-line results for VARs, systems integrators, and other reseller channels. At Digium, Jim is actively engaged in recruiting, en-abling, and growing the sales channel for both Aster-isk, Digium’s open-source telephony project, and for Switchvox, Digium’s premier UC telephony solution.
Juan Carlos FernandezCo-founder & COO, Telecomunicaciones Abiertas de Mexico SA de CV
Juan Carlos Fernandez serves as Co-founder and COO of Telecomunica-ciones Abiertas de Mexico, a Mexican company established in 2004 that
is devoted to bringing Asterisk and web solutions to SMB’s in Mexico. As Chief Operating Officer, he is re-sponsible for new product development and architec-ture. He has previously worked as a consultant for the United Nations in Kenya developing web applications
for different UN organizations like UNEP, UNON, UNDP and UNESCO. Education: Computer Engineering, ITAM University, Mexico, 1996, and a Master of Engineer-ing in Internetworking, Dalhousie University, Canada, 2002.
Justin HesterAsterisk Technical Trainer, Digium
Before joining Digium in 2013, Justin spent seven years performing various roles related to training and technical support in a large corporate contact center environment. Justin holds a BS from University of North Alabama.
Kayode AkintundeFounder and CEO, TheCodec Systems Limited
The founder and CEO of TheCodec Systems Limited. I Picked up a career in Voice Solutions at Resourcery Lim-ited, a leading ICT company in Nigeria,
haven completed a University degree in Computer Science with Mathematics. At Resourcery as a trainee engineer, I learn how to install, configure an deploy proprietary PBX systems. With that Telephony back-ground, my Linux skills, and Networking Knowledge, I discovered Asterisk in 2004, fell deeply in love with it, became an enthusiastic and fanatic of Asterisk by developing arrays of solutions from IP-PBX to Contact Centre Solution, and Customised IVR solutions
Keith CaseyDirector of Platform, OP3Nvoice
Keith Casey currently serves as Direc-tor of Platform for OP3Nvoice working to make APIs easier, more consistent, and help solve real world problems. Previously, as a developer evangelist
at Twilio, he worked to get good technology into the hands of good people to do great things. In his spare time, he works to build and support the Austin technol-ogy community, blogs occasionally at CaseySoftware.com and is completely fascinated by monkeys. Keith is also the author of “A Practical Approach to API Design”.
Kent BrooksDirector of IT, Casper College
IT Director Casper College since 2011
CTO/Dean of Distance Learning West-ern Oklahoma State College 14 years
Faculty Department Chair University of New Mexico Gallup 4 years.
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Lantre BarrChief Executive Officer, Blacc Spot Media, Inc.
Lantre Barr is not your typical techie given that he started in the world of architecture and construction. Tran-sitioning to the tech world in 2008,
Lantre began to tinker with the idea that technology is only confined by what the imagination can muster. In 2011, Lantre founded Blacc Spot Media with the objective to create cool and innovative solutions for enterprises and startups that were not afraid to be dis-ruptive. The introduction of WebRTC in 2011, created a unique opportunity to provide niche services using web real-time communications. To date, Blacc Spot Media has developed a wide range of applications from online therapy to online fitness platforms and is currently seeing 90% of all clients looking to integrate WebRTC solutions into their platforms or services.
Lefteris ZafirisLead VoIP Engineer, NTA Ltd
Passionate about Linux and commu-nications technology, working in the field of VoIP communications for about 10 years, mostly involved with free/open tools like Asterisk ranging from
embedded development to carrier grade deployments. Lately I’m focusing on the challenges of scaling Aster-isk based products in very big numbers of users. Active member of the Free software community, have been developing and sharing some of my code with the aim to contribute back and help the expansion of the Asterisk ecosystem. Fields of interest: Voice synthesis and recognition, Asterisk high availability, embedded Asterisk.
Luke DuquaineSoftware Developer, Schmooze Com, Inc.
Luke is the lead developer for the FreePBX EndPoint Manager and brings 20+ years IT experience supporting Windows, Unix, Novell and more. At Schmooze he also brings extensive
knowledge of infrastructure and hardware to the com-pany as well as helping design and build our Portal and other FreePBX modules.
Mark AmickDirector of Product Marketing, Digium
Responsible for being the unifying force behind Digium’s product strategy and execution, combining feedback from sales, marketing, development, customers and prospects to set the
strategic vision and lead the execution on new fea-tures and products. Direct a team of Product Managers in gathering and prioritizing product and customer requirements, defining the product vision, and work-ing closely with development to continue to deliver products that exceed customers expectations. Also work with sales, marketing and support to ensure our product supports our business model and is helping to grow Digium.
Martin KaplanHead of Engineering, Kaptec
Martin has 10+ year experience in IT an 5 years experience with Asterisk and commercial phone systems. In last 2 years he was heavily involved in building cloud multi tenant Asterisk
environment, integrating Asterisk with various web interfaces and building Asterisk around existing com-mercial PBX environment (Avaya). Martin has ability to combine his telephony, Contact Center and develop-ment skills, which allows him to quickly adapt in many projects.
Matt JordanEngineering Manager, Digium, Inc.
Matt Jordan is an Engineering Manager for the Open Source Software team at Digium, working on Asterisk. Matt joined the team in 2011, and since then has been involved in the devel-
opment of both Asterisk and the Asterisk Test Suite. His background in software development can best be described as “eclectic”, having worked in a variety of industries. Uniting the various experiences, however, is a firm belief in good software development practices and methodologies and the effect they have on pro-ducing quality software (and keeping software devel-opers from going insane).
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Matt RiddellManaging Director, VentureVoIP
Matt Riddell has been writing soft-ware since the age of six and work-ing with telephony software since around 1996. He started working with Asterisk in 2003 and started
the Daily Asterisk News in 2004. The Daily Asterisk News has since had over 2600 articles on various topics in the Asterisk community. He is the founder of VentureVoIP, a company that provide custom tele-phony solutions to businesses worldwide as well as consulting on various projects.
Melissa ShepherdAsterisk Training Manager, Digium
Melissa Shepherd is Digium’s Aster-isk Training Manager, having joined Digium in 2013 from CSC. She has over 15 years experience crafting and deploying technology solutions
and end-user training for government and non-profit organizations. Melissa holds an MLS from the Univer-sity of Alabama.
Michael Schloh von BennewitzComputer Scientist, Europalab Net-works
Michael Schloh von Bennewitz is a computer scientist specializing in network software, mobile com-puting, and client server design.
Responsible for research, development, and main-tenance of packages in several community software repositories, Michael actively contributes to the Opensource development community. Fluent in four languages, he speaks at technical events every year. Michael has presented for groups including Cable & Wireless, Nokia, Ubuntu, Mobile World Congress, and ClueCon.
Nir SimonovichCEO & Founder, Greenfield Technolo-gies Ltd
As an open source integration expert and evangelist, Nir Simionovich has delivered memorable conference speeches and participated in panels
in many parts of the world. These include North America, Central Europe, Eastern Europe, the Far East, Africa and, of course, Israel – where Nir is Based. He is a professional Asterisk application and platforms
develeoper with a flare for platform development in various fields. He is also an established author, hav-ing written two books about Asterisk.
Paul AshfieldSoftware Development Manager,Gage Networks
Paul has 20 years’ experience in application development, and is an enthusiastic and capable software development leader. He is particu-
larly focused on ease of use and is passionate about making the benefits of advanced technology easily accessible to all.
Paul has worked with Asterisk since 2006, develop-ing integrated application solutions. He leads the Touch Telephony development team, integrating Asterisk with a Web UI to control all aspects of call flow and device management.
Paul BelangerTechnical Lead, PolyBeacon, Inc
At PolyBeacon, Paul Belanger provides consulting services for IP Telephony, VoIP and Asterisk PBX. Specific interests include open source development for Asterisk,
OpenStack and Puppet.
Peter DunkleyTechnical Director, Acision
Peter is Technical Director at Acision. He graduated from The University of Edinburgh in 2000 with a BSc (Hons) in Computer Science. After gradua-tion Peter worked on a PSTN switch
developing signalling stacks for SS7, ISDN and simi-lar protocols and creating advanced routing and ser-vice applications. Since 2005 he has worked mainly with SIP first leading a team developing a PSTN gateway and then managing the development of a SIP Application Server. Peter joined Crocodile RCS as Technical Director in September 2010 before moving to Acision in November 2013 and has made numer-ous contributions to the Kamailio open source SIP Router project (particularly in the areas of presence, WebSocket, MSRP, and SIP Outbound) since then. Peter is one of the authors of the MSRP over Web-Socket draft (draft-pd-dispatch-msrp-websocket) and is a contributor to several open-source projects.
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Peter KellyCTO, Localphone/Voxbeam
Responsible for the design, deploy-ment and management of Voxbeam/Localphone’s international VoIP network including SIP origination, SIP to (Tier 1) PSTN termination and
IVR services. Previously at Ebuyer.com/Wrenkitchens.com was responsible for the technical design and implementation of stock, purchasing, semi-automat-ed WMS and manufacturing systems required for ebuyer.com and wrenkitchens.com. Led the success-ful development and deployment of these systems for Ebuyer from 2004-2008 and subsequent integra-tion and expansion of the systems for Wren Kitchens in 2009, including scope for a national retail store network. Also responsible for managing a team of developers based in the UK and USA.
Philippe LindheimerVice President, Schmooze Com Inc.
Among his many VP Executive func-tions, Philippe has been the Project Leader and primary Developer of FreePBX since 2006. Philippe’s inter-est in telecom began in the early
1990s when he worked at Hewlett Packard’s Europe-an Headquarters helping leading telephone compa-nies re architect their Business Support Systems. For almost three decades he has worked in the Engineer-ing industry in a range of technical consulting roles with Fortune 500 Companies and has helped grow FreePBX into the worlds most widely deployed Open Source PBX. Philippe has been involved in creating and presenting numerous technical training events on a worldwide basis and has spoken at many past Astricon developer conferences and Asterisk World. He is co founder of the popular Open Telephony Training Seminar series focused on real word Aster-isk/FreePBX technical and marketing/sales training.
Phillip MullisEntrepreneur, Consultant
Experienced entrepreneur, network engineer, integrator and architect, with a solid decade of asterisk experience and over 19 in telecom-munications, networking, and system
design. Phillip enjoys building many asterisk instal-lations from small to enterprise. His most recent installation is now hosting over 15,000 companies and growing rapidly.
Russell BryantPrincipal Software Engineer, Red Hat
Russell is a Principal Software Engineer at Red Hat where he works on the OpenStack project. He is an elected member of the OpenStack Technical Committee which pro-
vides overall technical oversight and governance for the OpenStack project. Russell served as the Open-Stack Compute (Nova) Project Technical Lead for two releases (Havana and Icehouse) and is still a member of the Nova Core team.
Russell also has a long history in the Asterisk com-munity. He has been contributing to Asterisk devel-opment in some capacity since 2004, including lead-ing the project for some time. Russell co-authored “Asterisk: The Definitive Guide” published by O’Reilly Media. He also contributed a chapter on Asterisk to the book “Architecture of Open Source Applications”.
Samuel Fortier-GalarneauSoftware Developer, Digium, Inc.
Samuel Fortier-Galarneau is a Soft-ware Developer at Digium, where he works on building real time commu-nications applications using Node.js and Asterisk as well as tools to en-
able developers to more easily develop applications using the new Asterisk ARI RESTful interface. Before joining Digium, Samuel worked on building web ap-plications in the real estate and telecom industry.
Sebastien CoutureVice President and CTO, Ubity Inc.
Sébastien Couture is Vice-President and Chief Tech-nology Officer (CTO) at Ubity, a cloud-based unified communications service provider. An expert in net-working and IP telephony, Sébastien is the main ar-chitect of a platform that now serves more than 500 businesses all over Canada and the United States.
Over the years, Sébastien has also worked on several projects that used telecommunications as a main tool for interactivity, such as “Legend of the Sea Lord” and “Awaz Do” from Indian artist Vishal Rawl-ley. More recently, Sébastien has lent his expertise to “Émilie”, an interactive web series that makes extensive use of telephony, text messaging as well as voice recognition to engage and interact with its viewers.
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Steve SokolDirector of Strategic Programs, Digium
As Director of Strategic Programs, Steve acts as an internal entrepreneur (sometimes called an Entrepreneur In Residence or Intrapreneur), working to identify and capitalize on emerging
technologies.
Susan ElderDirector of Marketing, Jenne, Inc.
Susan Elder serves as the Director of Marketing for Jenne, Inc., a value add-ed distributor of IP telephony, data, audio and video conferencing, and security technology products, includ-
ing equipment for the Enterprise and SMB markets. She has over 25 years experience leading marketing and communications for major organizations, includ-ing Invacare Corporation, a Fortune 500 manufactor of non-acute care medical produts, and the American Red Cross of Greater Cleveland as its Chief Market-ing Officer. Susan has lauched social media marketing initiatives for both the Red Cross and also for Jenne, Inc. Jenne now has a robust social media presence that has enhanced the company’s brand, image and reputation with customers and potential customers. Jenne’s LinkedIn Group, which Susan launched in February 2012, is a growing and dynamic group of CEOs/owners, head of sales, purchasing and other IT deicision-makers. She regularly consults with Jenne’s reseller customers on how they can launch social me-dia marketing initiatives for their own businesses.
Takahiko NagataEngineer, NTT Software
NTT is one of the largest companies in Japan. NTT Software is wholly owned by Nippon Telegraph and Telephone Corporation (NTT), providing solution services based on emerging technolo-
gies from NTT R&D laboratories. NTT Software start providing asterisk based solutions from around 2006 and already has several asterisk based products.
Hideyuki has been working on telecom area including ISDN/SIP/LTE/WebRTC etc.
Tim PantonDirector, Westhawk Ltd
Tim is a futurist with a strong prag-matic instinct.He takes new technolo-gies and combines them to create something that meets an unmet need. He does this by extensive prototyp-
ing and experimentation to see if a thing works and is desirable. Tim is currently looking at how to apply web technology, thinking and metrics to the entrenched voice communications market (aka the telephone). He has a bright red rotary dial phone on his desk to remind him how little things have improved in 100 years. Tim alternates between running startups and consulting for larger companies. He’s currently in a consulting phase. Past projects have included training simulation for oil platforms, time management of a space telescope and running a mobile phone network at the Burning Man festival.
Tom De MoorIT, Control Alt Dieliet
Tom De Moore has worked for more than 10 years in IT in a non-profit or-ganization. He has experience writing programs in PHP,JavaScript and Java.
Tony LewisCEO, Schmooze Com, Inc.
Tony comes from an entrepreneurial background, successfully starting and running various companies and
brings a wide background in aggressive sales and marketing To Schmooze Com and the FreePBX Project. His true passion is competing with FreePBX/Asterisk based PBX’s in a Legacy World and he has developed a successful program to give integrators an edge when selling against the Big Brand Competition. Mr. Lewis also has a vast knowledge of FreePBX and Asterisk and what their capabilities for customization bring to the table; and how to use the openness of Asterisk to help close sales.
Troy EgglestonCoordinator of Telecommunications, Clark County School District
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Exhibit Hall Map
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Premier SponsorDigium - Booth #19www.digium.comDigium, Inc. provides Asterisk software, telephony hardware, and on-premises and hosted Switchvox business phone systems that deliver enterprise-class Unified Communications (UC) and UC as a Service (UCaaS) solutions at an affordable price. Digium is the creator, primary developer and sponsor of the Aster-isk project, the world’s most widely used open source communications software. Asterisk turns an ordinary computer into a feature-rich communications server. A community of more than 80,000 developers and users worldwide uses Asterisk to create VoIP commu-nication solutions in more than 170 countries. Since 1999, Digium has empowered developers to create innovative communications solutions based on open standards and open source software, providing an alternative to proprietary phone systems. Digium Switchvox Cloud and other cloud-based products and services are offered through Digium Cloud Ser-vices, LLC, a wholly owned subsidiary of Digium, Inc. Digium’s business communications products are sold through a worldwide network of reseller partners.
Gold Sponsors
Schmooze - Booth #27/28www.schmoozecom.comWe offer a wide range of communication products ranging from Open Source FreePBX development and support to commercial PBX hardware, software and OEM platforms that allow both end users and system integrators from SOHO to Fortune 1000 companies to communicate. Vitelity - Booth #29/30www.vitelity.comWe understand you have a choice when it comes to VoIP providers and with more and more starting up every week it can make finding the right provider almost impossible. For almost ten years Vitelity has been providing VoIP, Fax, SMS, and hosting services to tens of thousands of customers worldwide. We have spent those ten years building the most reliable, feature-rich service offering available today.
Sponsors
Algo - Booth 16http://www.algosolutions.comAlgo’s innovative and cost effective SIP endpoints ease the migration path from legacy telephony to an Asterisk VoIP environment. SIP solutions include voice paging, loud ringing, visual alerting, notifica-tion, and security intercom. Visit the Algo booth to see the latest paging products, including the new 8188 Wideband G.722 SIP Ceiling Speaker.
allo.com - Booth #21http://allo.comALLO’s products are geared towards the SoHos / SMB segment and our hardware is solid, powerful and easy to configure thanks to well thought user inter-faces. Hardware is nothing without the software and user interface. ALLO’s solutions are an economical alternative to the major, pricy brands and in today’s economy that counts!
Aptus - Booth #22http://aptus.comLinking Asterisk with Unified Communication. FonB Base Package provides mobility and phonebook solutions to your Asterisk. It empowers users with the ability to: make, receive & transfer calls between their WebPhone, Mobile Phone and Desk Phone, sync their Google/iPhone personal Contacts. Users can also dial all their contacts easily including Asterisk defined Speed Dials and Extensions.
Avoxi - Booth #9http://www.avoxi.comAVOXI is a cloud communications provider focused on meeting the international calling needs of call centers, hotels, tour operators, and global enterprises looking to project a local presence. We provide tele-communications and enhanced VoIP solutions includ-ing toll free services, call recording, call termination, Cloud PBXs, call center solutions, toll free numbers worldwide, local numbers worldwide, SIP trunks, DID/DDI numbers, VPN solutions, business telephone sys-tems, VoIP phone systems, and conference calling.
Broadvox - Booth #5http://www.broadvox.comBroadvox is a leading domestic and international pro-vider of IP Communications. We enable our service provider partners and voice and SMS focused ISVs to succeed through cloud-based communications and applications.
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Cepstral - Booth 15www.cepstral.comCepstral LLC is a leading Text To Speech software company based in Pittsburgh , PA. Providing the Alli-son TTS voice and other speech technologies for the spoken delivery of information through Asterisk. Our high quality, natural sounding voices run on server, cloud, and even Raspberry Pi environments.
Elastix - Booth #24www.elastix.comElastix is an Open Source Software to establish Uni-fied Communications. Elastix’s goal is to incorporate all the communication alternatives, available at an enterprise level, into a unique solution.
Gage Networks - Booth #13http://www.gagenetworks.comGage Networks was established in 2002 and has developed long-term strategic partnerships with Vodafone, Polycom, BT, Equinix and PCCW to deliver global communications services across a private cloud. We pride ourselves on being at the forefront of technology with innovations enabling us to stay ahead of the curve. This drive results in our custom-ers being amongst the first companies to benefit from the latest technologies and solutions.
Grandstream - Booth #25www.grandstream.comGrandstream Networks, Inc. is an award-winning designer and manufacturer of innovative IP voice, video, and video surveillance products. Grand-stream’s products deliver superb sound and picture quality, rich telephony features, full compliance with industry standards, and broad interoperability with most service providers and third party SIP based VoIP products.
Jenne Distributing - Booth #23www.jenne.comJenne Inc., is a leading value-added distributor of business telephony, data, audio and video confer-encing, and security technology products for the SMB and Enterprise markets. Jenne is committed to providing installers, dealers and resellers with a broad product selection, competitive pricing, on-time accurate delivery, outstanding technical sup-port plus ongoing sales and technical training.
Kamailio - Booth #2http://www.kamailio.org/Kamailio® (former OpenSER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second.
QueueMetrics - Booth #20http://queuemetrics.com/Since 2004, Loway Switzerland develops complete, reliable and customizable call-center solutions.Our renowned QueueMetrics set up modern stan-dards in performance measurement, statistics and reporting. WombatDialer is a flexible, easy-to-use outbound dialing platform. Our mission is to put Swiss passion for precision and reliability at your service. Measure. Improve.
LumenVox - Booth #8www.lumenvox.comLumenVox has won numerous awards for innovation and technical excellence. Based on industry stan-dards, LumenVox’s core Speech Software is certified as one of the most accurate, natural sounding, and reliable solutions in the industry. The Speech Recog-nizer, Text-to-Speech Server, and Speech Tuner are the foundation of a successful speech solution. The LumenVox technology provides tools for you to ef-fectively connect and communicate with your users, increase user satisfaction, and improve employee productivity.
NETXUSA - Booth #7www.netxusa.comA true value-added distributor of industry lead-ing manufacturer products and recognized largest distribution partner of Digium products. NETXUSA supports more than 5,000 resellers and service pro-viders with our comprehensive solutions powered by ChannelWare®, that includes a new free service specifically for Digium Switchvox resellers, the Switchvox System Builder Wizard.
NTT Software - Booth #26http://www.nttsoft.comNTT Software Corporation was established in July 1985 when NTT became a private company after being a government organization. Our first busi-ness was designing and building of state-of-art and large-scale software for information communications for NTT Laboratories. Since then, NTT Software has developed a variety of products and technology; advanced free-dial development of a large-size com-mercial network service system, the DIPS Operating
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system maintenance and development service was transferred from NTT Laboratory, etc. Additionally, NTT Software provides network services, total system integration and system solutions.
OpenSIPS - Booth #1http://www.opensips.orgOpenSIPS is a multi-functional, multi-purpose signal-ing SIP server - it can act as SIP Router/Switch, SIP Registrar, Application Server, Redirect Server, Load Balancer / Dispatcher, Back-to-Back User Agent, Presence Server, IM Server, Session Border Controller, SIP Front-End, NAT traversal Server, IP Gateway (SMS, XMPP) and others.
OrecX - Booth #17www.orecx.comThe most affordable and easy-to-use call recording software on the planet. Companies record customer interactions today for quality assurance purposes, to resolve disputes and to comply with industry and/or governmental regulations. These same companies want a recording platform that they can count on and one that is affordable, easy to use and requires little to no maintenance. There is only one provider, however, that can prove it truly offers the most affordable, easy to install, maintain and use call re-cording software on the planet – designed primarily for contact centers, Asterisk users and ITSPs (Internet Telephony Service Providers). That company is OrecX.
Respoke - Booth #4www.respoke.ioRespoke makes it incredibly easy to add real-time communications capabilities to the things you build. Integrate text chat, audio calling, video collaboration, screen sharing, and peer-to-peer data with just a few lines of JavaScript.
Sangoma - Booth #11sangoma.comSangoma is a leading provider of hardware and soft-ware components that enable or en hance IP Commu-nications Systems for both telecommunications and data communications applications. Sangoma’s latest innovations and expanded product portfolio in clude Analog, Digital and GSM Interface Boards for Asterisk, SBCs, VoIP Gateways, Video MCUs, and other media processing products.
SimpleWan - Booth #12http://www.simplewan.comSimpleWan™ is a subscription and hardware based firewall designed for small, medium and enterprise level businesses that allows them to improve over-all connectivity. In a nutshell a SimpleWan™ is a wholesale product for service providers that deploys network equipment of any kind on their customer’s premises. Our goal is to provide reliability, and the tools to troubleshoot broadband technology quickly in realtime when services go down. We have many unique add-on services that lower the cost to man-age hosted subscribers. Our mission is to bring all these tools and services at a wholesale level to content and service providers and lower their overall customer churn.
TeleDynamics - Booth #14www.teledynamics.comTeleDynamics, established in 1981, is a wholesale supply house and fulfillment company based in Austin, Texas. They stock products from more than 150 vendors including their leading brand of Panasonic. Items in inventory include everything from telephone accessories, cordless telephones, general audio products, personal care items, outdoor gear, computer/telephony products, managed rout-ers and data switches to the most advanced digital business phone systems, SIP phones and IP phone systems on the market today.
Vestec - Booth #10https://www.vestec.comVestec is changing the game in conversational speech technology. Leveraging the power of Artifi-cial Intelligence, Vestec’s mission is to “popularize” speech by removing the obstacles that have tradi-tionally held natural language applications back-- unleashing the simplicity, versatility and affordability of natural speech interaction with machines and devices.
Xorcom - Booth #18www.xorcom.comXorcom designs and manufactures business telepho-ny solutions that support both traditional PSTN and VoIP. All products are based on Asterisk, making them easy to install and maintain, and cost-effective. Xor-com utilities maximize uptime: backup and restore, dedicated redundant power supplies, TwinStar™ – a unique hot failover solution for the entire PBX.
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Asterisk Essentials | Learn Asterisk online at your own pace.
This video course is designed to rapidly guide a new user through the installa-tion and basic configuration of Asterisk. Key telephony concepts are introduced, explained, and implemented. The course is heavily example-based, with a focus on the practical knowledge required to successfully administer an Asterisk system. Though some time is given to architecture and theory, we’ve only included the most helpful information. We’ve left out the non-essentials so that you can get up to speed with Asterisk as quickly as possible. Best practices are described through-out, and many examples are supported with Asterisk documentation.
A FREE Video Sample and Course Outline are available online at: www.digium.com/en/training/asterisk/essentials
Asterisk Fast Start | Get up to speed quickly on Asterisk!
Asterisk Fast Start is a three-day hands-on course, designed to equip you with the knowledge and skills necessary for an entry-level Asterisk Administrator to be suc-cessful on the job. Using a combination of information-rich lectures and practical lab exercises, this course familiarizes you with Asterisk and the environment in which it operates, both in terms of operating system and telephony (traditional and IP) connections. This course will take you from a freshly installed Linux PC through to a fully configured and working Asterisk implementation. You do not need to have previous experience with Linux, telephony or Asterisk.
Course Outline and Schedule is available online at: www.digium.com/en/training/asterisk/fast-start
Asterisk Advanced | Expand your knowledge of Asterisk.
The Asterisk Advanced course is a five-day hands-on course, that covers the knowl-edge and skills an advanced Asterisk administrator should know to be effective at his or her job. It is a combination of both lectures and labs designed to give the students both the theory behind the concepts and the hands-on experience to be able to effectively deploy Asterisk-based solutions. This course is the successor to the wildly popular Asterisk Bootcamp, which was retired in 2008. In Asterisk Advanced, students will learn to create advanced dial plans and innovative tele-phony solutions by using the features of Asterisk. Along the way, students will also get important insight into VoIP. Students will also explore connecting Asterisk to a variety of devices, including analog phones and phone lines, digital phone lines, VoIP phones and VoIP providers.
Browse the Course Topics and Schedule online at: www.digium.com/en/training/asterisk/advanced
Upcoming Course Locations Include:
Dubna, RussiaOxford, UK
Huntsville, AL, USMadrid, Spain
Toronto, CACape Town, ZA
Upcoming Course Locations Include:
Dubna, RussiaOxford, UK
Huntsville, AL, USMadrid, Spain
Toronto, CACape Town, ZA
Asterisk TrainingWhether you want to learn a little more about Asterisk or become certified, Digium offers a variety of online and instructor-led training to meet your needs.
FREE VIDEO SAMPLE
Get a feel for our online course format in this free
sample on SIP Callflow
Free Asterisk Essentials Videos - Phone Systems...http://www.digium.com/en/training/asterisk/esse...
http://kaywa.me/8R0iq
Download the Kaywa QR Code Reader (App Store &Android Market) and scan your code!
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What type of products are listed? AsteriskExchange™ listings include hardware and software products and components, including: IP phones, soft phones, edge devices, gateways, call center applications, voice prompts, text to speech and speech recognition software, complete PBX systems, gateway hardware and much more.
What are the advantages of being listed? AsteriskExchange™ is the premier marketplace for all things Asterisk. With Asterisk developers and millions of Asterisk users and downloads worldwide, AsteriskExchange™ provides high visibility and drives targeted traffic to your product.
Will products be reviewed and ranked by user popularity or downloads?Yes, AsteriskExchange™ provide provides ranking services, so the most popular and highest rated products will premium listingreceive priority in the listings. Users also will be able to post narrative reviews, sharing opinions and viewpoints from around the globe.
How do I list my company’s product?Digium reserves the right to determine which products may be listed. Submit your product(s) using the online product listing wizard. Your submissions will be reviewed and if approved, you will receive a price quote and instructions for activating your listings.
My application is not a free open source tool and I am not prepared to share source code. Can I still list my product on AsteriskExchange™?Yes. There is absolutely no requirement that you make your source open. Many software products listed on AsteriskExchange™ are proprietary, binary files -- not open source.
I would like to have my product certified by Digium and promote this product on AsteriskExchange™. What is the process?Digium offers exclusive product testing and certification for products that interoperate with Asterisk. Certification is a separate process from listing your product on the AsteriskExchange™, but certified products do receive special billing and users can filter to view only certified products.
AsteriskExchange AsteriskExchange™ is the online marketplace for products built on or integrated with Asterisk.www.asteriskexchange.com
Asterisk CertificationThe Digium-certified Asterisk Professional (dCAP) certification is a verification of your knowledge of Asterisk. The certification covers a specific released, stable version of Asterisk. The program is offered at the end of the Asterisk Advanced training course. The dCAP test consists of a 115-question written exam concerning Asterisk and Aster-isk-related technology, and a hands-on practical lab exam in which you are asked to configure a PBX according to a given specification.
The Digium Certified Asterisk Administrator (dCAA) program is intended to validate ba-sic competency with Asterisk. It requires general telephony and VoIP knowledge as well as specific Asterisk knowledge related to installing, configuring, and administering an Asterisk system. dCAA certification lasts 2 years and is awarded after earning a passing score on the dCAA examination. The dCAA exam is administered online and consists of 60 multiple-choice questions testing your knowledge of Asterisk, VoIP, and telephony.
To enroll for certification in North America, locate a dCAP course in the Training Course Locator online. at www.digium.com/training. Global authorized training partners also offer the dCAP at the end of each Asterisk Advanced Course, contact the partner directly for details. To take the certification while at AstriCon, please visit the AstriCon registration desk for detailed information.
