Asterisk SCF is a framework that will allow developers to move beyond voice and also create open source video and text that can be used for anything from embedded applications to enterprise and carrier solutions, according to Digium. It is delivered as a system of distributed components that can be deployed in clusters on a single system or on many systems, and will support pretty much all real-time IP communications including video, wideband audio, chat and desktop sharing.Asterisk SCF seeks to meet four requirements for businesses looking to deploy open source communications solutions:

• Performance
• Scalability
• Fault tolerance
• Extensibility

Live from Astricon (well it was when we recorded it), Kevin Fleming, director of software technologies at Digium joins us at talk about the big Digium announcement.  This discussion was made possible  by Tim Panton and Michael Graves. Kevin, Tim and Joshua Colp were in a room, surrounded by John Todd’s tie-dyed shirts.

Thanks to those who requested this on our mailing list! We have liftoff! (VoIP Security)

VoIP Abuse Project . J. Oquendo

“Arkeos is a program primarily used on Unix based servers running the Asterisk Open Source PBX. The application mimics a valid extension in which an attacker, after bruteforcing an account, attempts to place calls. What the attacker doesn’t know is that their calls go nowhere. The sole purpose of allowing them to perceive they have an account is to track them. Where they come from, what accounts they use, what VoIP account scanners they use, what numbers they try to dial. Because of the flexibility of Asterisk and Unix as a whole, the application does a lot more than stated however, I will not disclose too much otherwise any attackers reading this might catch on. Then I won’t have anyone to play with.”

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.

Allison Smith (@VoiceGal) pinged me about the collection she recorded for Joey Lindstrom for use with Asterisk pbx. The full set, which Mr. Lindstrom has generously decided to place in the public domain is available for dowload free here:
http://vuc.li/FunnyAllison.

Joey wrote to say this was his way to give something back to the community, and he’s asking Digium to put these prompts in the Asterisk distribution should they wish to do so.

2600hz is home to a collection of open-source telephony software that enables the use of the FreeSWITCH, Asterisk and YATE switching libraries. Initially built around the blue.box project, we aim to provide a collection of software to power your GUI, your cloud-based telephony switch and/or your monitoring and maintenance tool set.

Video testing – Next week we’ll be doing a real video conference, watch the site for more info

ClueCon - No one wished to comment. Whatever.

OpenVBX is a web-based open source phone system for business.
With a Twilio account and a web server with PHP 5.2+ and MySQL 5, you can build your own “hosted pbx”.

Build your own custom phone applets with just a little bit of PHP. Rebrand and resell OpenVBX to your customers.

Give every user their own phone number and personal conference line. Dial whole departments, share voicemail messages with the team. OpenVBX is for companies and collaboration.

Adam Ballai, lead engineer on the OpenVBX project and Twilio CEO Jeff Lawson are our guests.

Both SIPVicious and Adhearsion have announcements to make. Sandro joins us to explain “svcrash”, a new addition to the tool kit to stop attacks when they happen:

A new tool has been added to SIPVicious – svcrash.py. As the name implies, it crashes something – svwar.py and svcrack.py. This tool is meant to be used by system administrators and organizations that are receiving unauthorized scans on their exposed IP PBX.

Ben joins us to give some details about the evolution of Adhearsion which has had a recent version release:

This release marks the continuation of the platform with some major enhancements and bugfixes:

• Support for Asterisk 1.6 (continued backward compatibility with Asterisk 1.4)
• Added support for ActiveLDAP
• ActiveRecord now works properly with Adhearsion components
• Daemonizing of Adhearsion no longer truncates log files
• Escape commands sent to Asterisk via AGI
• Asterisk Manager Interface (AMI) events now work when daemonized
• Various enhancements for improved Asterisk 1.6 support
• Various bugfixes

The Incredible PBX: Free Skype, free worldwide SIP, free U.S./Canada PSTN plus rock-solid Asterisk security using the new Zero Internet Footprint™ design.

In addition to all of the Orgasmatron magic including free calling in the U.S. and Canada courtesy of Google Voice, you now get some terrific new features: randomly generated passwords for all of your extensions, free Skype support and a new backup module both of which we’ll introduce over the next few weeks. And CallerID Superfecta now is preconfigured to work out of the box with support from dozens of providers worldwide.

Martin Sunstrum, the founder of Aksys Networks joins us for a presentation of the KONNECT line of business VoIP phones.

He will give an overview of the KONNECT Business Phones and explore the tight feature integration with Asterisk-based PBX systems.

Lastly, he will present the intriguing KONNECT Office Phone System, a multi-site office phone system that requires no PBX or hosted service.

[audio:http://recordings.talkshoe.com/TC-22622/TS-338121.mp3|titles=Sam Reports on Voipathon]

Mihaela from Token d.o.o. joins us for a chat about Sam Reports, custom software from her family company located in Croatia. She and her husband have been running an Asterisk integration shop since 2007 and writing custom software to ease and enhance working with Asterisk.

SAM Reports is aimed at small businesses using Asterisk. I built it to solve the pain that I had : provide my clients, small businesses, with a reporting solution without the need to make any changes to their Asterisk boxes. And also to give them reporting that’s easy for them to use. I wanted to be able to offer it to both my call-center clients and those without, therefore the application handles both CDRs and queues. SAM Reports enables the end users of Asterisk solutions to have rich and powerful reports without the hassle of installing additional software.