From Jason Goecke’s blog post – read the whole post here on the Tropo blog. Jason made the announcement to the world about this new project on the VoIP Users Conference!

“We have decided to take it a step further and created a Tropo Scripting application that turns Tropo into a giant Asterisk application platform in the cloud.   You can now run just about any Asterisk AGI application on Tropo.

Tropo AGItate was started on the Nerd Bird (good to have in-flight WiFi) from San Jose to Austin, on my way to LoneStar Ruby Conference. Jim Freeze – the organizer of LSRC – had recently been to AdhearsionConf in San Francisco; I wanted to be able to show something extra special during my talk there. On that one flight, I was able to get the basics working and show Tropo emitting AGI during my talk, just like that. (For those non-Asterisk folks out there, AGI is an API that lets external applications connect in to Asterisk and fully control it).”

Key points:

• Tropo can now speak AGI over TCP to any FastAGI server, including Adhearsion, PHPAGI, Asterisk-Java, etc.
• A long list of Asterisk commands are supported, and of course all the Tropo ones.
• While the script is written in Ruby, no Ruby knowledge is necessary to use it. Just point your Tropo application to the Ruby script on Github, upload a configuration file via FTP or WebDAV to your Tropo account, and you are ready.
• Full support of Tropo Speech-Synthesis (TTS) and Speech-Recognition (ASR), but also for Asterisk sound files. Yes, tt-monkeys works!
• Fail over to a SIP URI, in case your FastAGI server does not respond. This could be to another Tropo application, another Asterisk box, or anything that supports SIP.
• You get all of the Tropo channels over AGI, including SMS, Instant Messaging and Twitter, all using the same application.
• Support for custom SIP Headers in and out of Tropo.

Asterisk 1.4 the Professionals Guide by Colman Carpenter, David Duffett, Nik Middleton, and Ian Plain
One or more of the authors of this recent book on Asterisk will be with us to talk about a variety of related topics. This book represents the wisdom and thoughts of front line consultants. The knowledge they impart will prove informative, thought provoking and be of lasting interest to Asterisk professionals.

I’ve had a chance to look through the book, and I’d recommend that anyone who is serious about Asterisk read it as it addresses a lot of areas that have rarely been discussed in a single book.

Considerations about connecting to cellular networks, reflections on knowledge needed for running your own consulting business are just a couple of examples a new material I’ve not seen elsewhere.

Grabbing the list from one of the sites about the book, here are some examples of what’s to be found there:

What you will learn from this book?

* Master advanced dial plans, call routing considerations, and speech technologies.
* Learn all about AGI, AMI, ENUM, and DUNDi.
* Discover how to build an enterprise-scale Asterisk-based solution for mission-critical applications.
* Discover the DAHDI framework, and the way it replaces Zaptel.
* Understand how to apply localization and customization techniques in different territories.
* Learn how to interface Asterisk with analog and digital telephony systems, as well as wireless technologies.
* Appreciate what you need to do as an Asterisk professional when assessing customer needs, Quality of Service, pitching for business, and customer support.
* Understand best practices through example code and detailed commentary.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-263999.mp3]

Great call, thanks to David and Nik!

IRC Transcript VUC 2009-09-25 The boys do shine in these quizzes!

Note: From http://voip-info.org “m: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option ‘r’, obviously. Use m(class) to specify a class for the music on hold. “

Question: Dial()'m' argument
[7:18pm]Ottone: music on hold for dialtone
[7:18pm] kfife: music on hold class
[7:18pm] ricko73: moh?
[7:18pm] ssokol: music class
[7:18pm] Ottone: it is for "ringing tone"
[7:19pm] J-MN: shouldn't it be the looser gets the book,
    the winner doesn't need it but the looser does...

Port for AMI?
[7:23pm] _3vain: 5038
[7:23pm] ssokol: 5038
[7:23pm] ricko73: 42 <- heh
[7:23pm] Ottone: 5038

[audio:http://voipusersconference.org/wp-content/uploads/NirSim.mp3|titles=Nir Simionovich on Asterisk AGI programming]

I had the pleasure of meeting one of the more vibrant and outgoing characters at the conference and recorded this interview about his latest PHP AGI book. I only wish I could have shared the stories, jokes and general mirth happening in the traditional German restaurant when Tim Panton, Jason Goeke, Nir and I shared a meal.

Nir’s presentation at AMOOCON was about running Asterisk under EC2 and I hope to get him to talk about this some day as he has an actual large-scale case he deployed in this way rather than just a theoretical one.

Nir’s book, Asterisk Gateway Interface 1.4 and 1.6 Programming is a great introduction to serious programming using the AGI and PHP. The book takes you through some good solid advice for truly professional programmers on how to do stable programming in the Asterisk AGI environment. This is the book you need if you are trying to do anything serious with Asterisk in PHP.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-198844.mp3]

IRC Transcript 2009-03-27

SIP hacking has escalated and there was a wave of it in recent times. John Todd published a simple common sense set of rules that are worth reading. One of the main risk areas is using user names and secrets that are too easy to guess (such as extension 200, username=2000,password=wakeme).

More on SIP for Skype etc. That looks like it will be an ongiung discussion. See the session with Michael Robertson above.

We briefly reviewed a new book called Asterisk Gateway Interface 1.4 and 1.6 Programming. John Todd and /me have both been looking at it and we both think it looks good. Since the discussion, I’ve read more of it and I will talk more about it in a future session.

Amazon link (beer money!)
Asterisk Gateway Interface 1.4 and 1.6 Programming

We also mentioned AMOOCON where we can meet up for those of you who will be in Europe or are already there. I will be presenting two papers, one on VoicePHP and one on the VoIP community and various social networking tools.