RIP, Skype for Asterisk

Digium announced today the official end of Skype for Asterisk– ending anyone’s dream of a more friendly, open, Skype under Microsoft.

Their email, stated:

We expect that users of Skype for Asterisk will be able to continue using their Asterisk systems on the Skype network until at least July 26, 2013. Skype may extend this at their discretion.

The announcement of Skype and Asterisk came during Astricon 2008, a little less than 3 years ago. After almost a year, a beta program was announced with a pay per use licensing announced shortly thereafter.

The full announcement:

Product notification:

Skype for Asterisk will not be available for sale or activation after July 26, 2011.

Skype for Asterisk was developed by Digium in cooperation with Skype. It includes proprietary software from Skype that allows Asterisk to join the Skype network as a native client. Skype has decided not to renew the agreement that permits us to package this proprietary software. Therefore Skype for Asterisk sales and activations will cease on July 26, 2011.

This change should not affect any existing users of Skype for Asterisk. Representatives of Skype have assured us that they will continue to support and maintain the Skype for Asterisk software for a period of two years thereafter, as specified in the agreement with Digium. We expect that users of Skype for Asterisk will be able to continue using their Asterisk systems on the Skype network until at least July 26, 2013. Skype may extend this at their discretion.

Skype for Asterisk remains for sale and activation until July 26, 2011. Please complete any purchases and activations before that date.

Thank you for your business.

Digium Product Management

Anyway, Leif posted a great article on his “Asterisk, and other worldly endeavours” blog today regarding ENUMplus and ISNs. Check the article on his blog: Musings about ENUMplus and ISNs.

The Asterisk Development Team has announced security releases for the following
versions of Asterisk:

* 1.2.40
* 1.4.29.1
* 1.6.0.24
* 1.6.1.16
* 1.6.2.4

These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The releases of Asterisk 1.2.40, 1.4.29.1, 1.6.0.24, 1.6.1.16, and 1.6.2.4
include documention describing a possible dialplan string injection with common
usage of the ${EXTEN} (and other expansion variables). The issue and resolution
are described in the AST-2010-002 security advisory.

If you have a channel technology which can accept characters other than numbers
and letters (such as SIP) it may be possible to craft an INVITE which sends data
such as 300&Zap/g1/4165551212 which would create an additional outgoing channel
leg that was not originally intended by the dialplan programmer.

Please note that this is not limited to an specific protocol or the Dial()
application.

The expansion of variables into programmatically-interpreted strings is a common
behavior in many script or script-like languages, Asterisk included. The ability
for a variable to directly replace components of a command is a feature, not a
bug – that is the entire point of string expansion.

However, it is often the case due to expediency or design misunderstanding that
a developer will not examine and filter string data from external sources before
passing it into potentially harmful areas of their dialplan.

With the flexibility of the design of Asterisk come these risks if the dialplan
designer is not suitably cautious as to how foreign data is allowed to enter the
system unchecked.

This security release is intended to raise awareness of how it is possible to
insert malicious strings into dialplans, and to advise developers to read the
best practices documents so that they may easily avoid these dangers.

For more information about the details of this vulnerability, please read the
security advisory AST-2010-002, which was released at the same time as this
announcement.

Asterisk 1.2.40 also contains a backported dialplan function called FILTER() in
order to allow the filtering of strings as described in the best practices
document.

It should also be noted that the 1.6.x series of Asterisk had release candidates
available as versions 1.6.0.23-rc2, 1.6.1.15-rc2, and 1.6.2.3-rc2. These will
either be released as 1.6.0.25, 1.6.1.17, and 1.6.2.5, or if another round of
RC changes is necessary, those versions numbers will be used with -rc1 appended.

For a full list of changes in the current releases, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.2.40
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.29.1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.24
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.16
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.4

Security advisory AST-2010-002 is available at:

http://downloads.asterisk.org/pub/security/AST-2010-002.pdf

The README-SERIOUSLY.bestpractices.txt document is available in the top-level
directory of your Asterisk sources, or available in all Asterisk branches from
1.2 and up.

http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt

Thank you for your continued support of Asterisk!

Team Forrest describes how to install, configure, and then email your fax as a pdf. Check out the post at Team Forrest.

http://www.pindropsoup.com/2009/10/astricon-update.html

Now that we’ve returned from the Phoenix sun, we’ll probably write our own summary in the future.

Ty would be proud...

Are you ready? Tomorrow, Asterisk fans, fanatics, developers, users, and more will gather in Glendale, Arizona for AstriCon 2009. AstriCon, the official conference for Asterisk, runs from October 13 – 15.

This year, yours truly will speak (and very honored to do so) in a talk titled “Asterisk Applications – Unexpected Hurdles.”  (summary below)

But wait, there’s more…

Just in time for AstriCon, Digium pulled a Ty Pennington on the ol’ Asterisk website. It looks amazing. Just a great way to start the conference!

I’m extremely excited to be heading West and cannot wait to share the information learned. Get ready for a great show!

About Asterisk

Asterisk is free, open source software provided under the GNU General Public License (GPL). Asterisk is the most popular open source software available, with the Asterisk Community being the top influencer in VoIP.

Why free? It’s just how Digium rolls. They really take that GPL open source to heart.

For more information, please check out:

• Asterisk.org
• Digium, Inc.
• AstriCon.net

Summary of Fred’s talk:

With Asterisk AGI programming, almost anything is possible. From phone based payment systems to providing real-time information, Asterisk makes it possible to bring information to anyone with a phone. Sometimes, even the simplest applications can have unexpected consequences. Building a real-time Parking Garage availability application in Ann Arbor, Michigan was met with great appreciation by residents but blocked by government who didn’t understand how Asterisk gathered data — interpreting it instead as a Security risk. The talk would explain that when building even the simplest public application, the designers should be familiar with public access laws and be able to articulate how their application gathers data. Freedom of Information will also be discussed.

When Allison Smith Talks IVR, You Should Listen

Through her blog, Allison recently wrote about the top ten mistakes consistently made in IVRs. It’s a quick read. It’s a good read. And, she’s got enough street cred that when she says it’s an IVR mistake, you should just say, “Thank you ma’am, may I have another.”

And when you’re done reading the Top Ten IVR Mistakes, check out The IVR Voice as well.

Skype for Asterisk Beta is a download that lets you integrate your Asterisk system with the Skype network.

With Skype for Asterisk, you can:

• Make Skype to Skype calls
• Call landlines, cellphones, even grandma!
• Receive SkypeIn calls
• Make multiple Skype calls simultaneously using the same Skype account
• Read Skype profile fields
• Support DTMF
• Set and retrieve online status
• Handle incoming Skype calls using your dialplan
• Use the Asterisk PBX for voice and the Desktop for IM
• And much more!

And you can do this all from your Asterisk PBX!

Ordinary Skype is a mess. You need your desktop and some sort of sound equipment just to make a call. Crazy. Skype for Asterisk Beta has the muscle to use your phone system directly with the Skype network. Use Skype for Asterisk Beta to provide an IVR or Sales portal for your company. You can use Skype for Asterisk Beta at home while watching TV. You can even use Skype for Asterisk Beta to send your Skype calls to one central voicemail.

Whether your Skype needs are large or small, Skype for Asterisk Beta can handle it all. Skype for Asterisk Beta is free, so it pays for itself.

Through Digium’s exclusive Beta offer you can download Skype for Asterisk Beta at the very low price of Free.

But wait, there’s more!

Skype for Asterisk Beta is a limited time offer. You have to act now to download and register the software. Skype for Asterisk Beta can only be downloaded until August 7th and used until August 31st. And of course, being beta, there’s some betaness to contend with.

So act now and download the Skype for Asterisk Beta software today directly from Digium.

(In case you hadn’t guessed, this is also our homage to Billy Mays.)

Here’s what Digium’s John Todd posted:

I know many of you have been waiting for this for a while, so I’ll  keep this short:  The Skype for Asterisk Public Beta is now available on the Digium store.

We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium’s web store and purchase (for zero dollars) the Skype For Asterisk product. The web store does require a Digium.com account, which can be set up during the purchase process if you don’t already have one. Once the web store process is complete, you will be e-mailed your license key and directions on where to download Skype For Asterisk beta software.

This is a “time-expiring” beta – the software will stop working on August 31.  The download is also currently time-limited – it will be available until August 7 on our website.  After the 31st, you would need to have purchased a license for the SfA software (sorry, no pricing that I can give you right now – that will be a separate announcement.  I’m just the community guy – I have no idea about pricing or commercial contracts or the like, so please wait until that’s been announced as I will find out about the same time as you do.

Trial “purchase” page:
http://store.digium.com/productview.php?product_code=804-00019

JT

While you’re downloading Skype for Asterisk, read Michael Grave’s post over at Graves On SOHO VoIP.

When I first started working with VoIP, I began to hate telephony, and any and all things telephone related. This bothered me on many levels. You see, as a kid, I loved telephones. Growing up in the “big city,” pay phones seemed to be on every corner. Family stories talk about walking several blocks extra, just to avoid me seeing and wanting to play with a phone. But, as usual, I digress…

When I worked with an unnamed switch (let’s just say it rhymed with Broadmoft), I hated working with VoIP. I knew there had to be a better way and started playing with Asterisk. Soon, my memories of playing with phones started coming back and my love rekindled. Now, I look forward to working with phone systems, only because I truly feel that the use of a phone can only be limited by your imagination. And with companies like Twilio, Adhearsion, and Digium, the community of telephone developers seems only to grow.

With that long winded introduction, let’s discuss today’s topic — telemarketer torture. I, like every other person I know, receives the random yet continual undesired telemarketer call (yes, I’m on the DND databases and let’s not get into that). Thanks to Asterisk, I can send my telemarketers to a little place I call the Annoyatron.

Now, many developers and users implement their own version of Telemarketer Torture. Some like using IVR’s. Some like endless ringing. Personally, I like to keep them on the line for a long time. You see, since I add numbers to the Annoyatron after they call me, by the time they reach the Annoyatron they have already called and wasted my time at least once before. So, instead of just having them hang up and move on to the next home, I like to see if I can keep them talking for a while. My Goal? At least 2 minutes.

I use Asterisk’s “WaitForSilence” command to keep my torture conversational. When there’s a pause, the Annoyatron will play a file. While the telemarketer speaks, the Annoyatron will patiently wait. You put it all together, and wala — the Annoyatron Telemarketer Torture.

Today, I received unwanted calls regarding long distance to India. I added the number to the Annoyatron and well, the results of their continued calls no longer annoy me. Here are two examples:

Listen to Call 1:
Download audio file (annoyatron.mp3)
(or you can download the MP3)

Listen to Call 2:
Download audio file (annoyatron2.mp3)
(or you can download this MP3, too)

Ok, so here’s an example of how you would write the dialplan in Asterisk:

[annoyatron]
exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,Playback(annoy/annoy-hello)
exten => s,n,WaitForSilence(2200)
;...
; record a file for "your side" of the conversation
; wait for silence, and then play it
; lather rinse repeat
;...
exten => s,n,Hangup()

Simple, no? Just one of the reasons Asterisk allowed me to enjoy working with telephones. Awwww.

We’d love to hear about your fun examples with Asterisk. And you know, Leif Madsen is requesting some ideas for Asterisk Recipes himself.

Asterisk is free, open source software provided under the GNU General Public License (GPL). Asterisk is the most popular open source software available, with the Asterisk Community being the top influencer in VoIP.

Why free? It’s just how Digium rolls. They really take that GPL open source to heart.