Posts categorized "Asterisk"

Keynote at AstriCon on Oct 14: Open Source And The Global Disruption Of Telecom - What Choices Will We Make?

Astricon danyork 660px

Two weeks from today I'll be in Orlando giving the opening keynote address at AstriCon 2015. The abstract of the session is:

Open Source And The Global Disruption Of Telecom - What Choices Will We Make?

Wednesday, October 14th, 2015 - 9:00 am to 9:45 am - Pacifica Ballroom 7

There is a battle raging for the global future of telecommunications and the Internet. Taking place in networks, board rooms and legislatures, the battle will determine how we all communicate and what opportunities will exist. Will telecom support innovation? Will it be accessible to all? Will it give us the level of security and privacy we need to have the open, trusted Internet? Or will it be restricted and limited by corporate or government gatekeepers?

The rise of voice-over-IP has fundamentally disrupted the massive global telecommunications industry, infrastructure and policies. Open source software such as Asterisk has been a huge driver of that disruption and innovation.. but now what? What role do platforms such as Asterisk play in this space? And what can be their role in a telecom infrastructure that is now mobile, increasingly embedded (Internet of Things) and more and more using proprietary walled gardens of communication?

Join the Internet Society's Dan York in an exploration of what the future holds for telecom infrastructure and policy - and how the choices we make will determine that future.

Sounds great, eh?

Now I just have to deliver on that lofty rhetoric! :-)

Seriously, though, I'm very much looking forward to giving this presentation and I'm delighted that the folks at Digium asked me to speak. We're at a critical time in the evolution of our global communications infrastructure... with everything moving to IP and also moving to mobile, there are incredibly important choices we have to make for our future.

In the talk, I'll be speaking about the scenarios we have for what our future Internet could look like. I'll be talking about the role of open source. I'll be challenging the audience with some questions to ponder. I'll touch on some of the incredibly important - yet hard to understand - global policy issues such as the upcoming WSIS+10 Review in December - and why an open source developer should even remotely care! I'll of course hit on security issues and the rise of mobile... and more...

I'm excited!

I'm also excited to finally attend an AstriCon event. I used to write about Asterisk a good bit and for a while was running my own server in my home office for VoIP... but in all that time I never was able to work in attending an AstriCon!

If you are going to be there in Orlando, please do say hello! (There's still time to register!)

P.S. And yes, Olle Johansson, I'll be sure to work in at least one reference to IPv6! And TLS, too! Don't worry! :-)

Are They Crazy? Digium Enters The Phone Game With Asterisk IP Phones

DigiumphonesWhen I first saw the news today, my immediate reaction was:
Seriously? Digium is coming out with phones???
In a rather fascinating move in an already extremely crowded market, Digium announced today that they will be producing "Digium Phones", a new line of IP phones specifically targeted at users of Asterisk and Switchvox (both Digium products). They tout among the benefits:
  • Crystal clear HD Voice
  • Simple setup and installation
  • Tightest integration with Asterisk
  • Built-in & custom applications
  • A built-in "app engine" JavaScript API

There will be three models available:

  • D40—An entry-level HD IP phone with 2-line keys. Priced at $149.
  • D50—A mid-level HD IP phone with 4-line keys and 10 quick dial/BLF keys with paper labels. Priced at $179.
  • D70—An executive-level HD IP phone with 6-line keys and 10 quick dial/BLF keys on an additional LCD screen. Priced at $279

The news release indicates they will be available in April and are currently on display at ITEXPO this week down in Miami. A datasheet is available

Application Platform

What is perhaps most interesting to me is the "app engine" included in the phone. From the news release:

Digium phones include an app engine with a simple yet powerful JavaScript API that lets programmers create custom apps that run on the phones. They aren’t simply XML pages; Digium phone apps can interface directly with core phone features.

Many IP phone vendors have tried various systems like this to let developers build more apps into the phone with varying degrees of success. What makes Digium different, though, is that it comes from the developer community. The history of people working with Asterisk is the history of tinkering and hacking away on the systems. In fact, in the early days, that was all you could do. No fancy GUIs... just configuration files and cryptic APIs. As a result, Digium has a very strong developer community (they claim 80,000+ developers) who just may be able to make use of this new API.

What remains to be seen is what kind of applications you can really build with these phones - and how easy it is to install and or use these apps.

Are They Crazy?

But are they crazy for entering the already insanely-crowded IP phone market? Particularly at a time when enterprise smartphone usage is increasing - and may often be the preferred communication medium? And when people are becoming increasingly comfortable with softphones, courtesy largely of Skype and "Unified Communications" desktop apps like Microsoft Lync and similar apps from Cisco, Avaya, Siemens, IBM and more?

I completely understand that Digium would want to make the Asterisk "user experience" much easier and simpler. Particularly as Digium continually seeks to move beyond their traditional more developer-centric audience into businesses and enterprises. Many of those folks want a system that "just works." If they can order a system from Cisco or Avaya that comes complete with the IP PBX, IP Phones, etc. and it all just works, they may choose that over a less-expensive but harder-to-put-together solution using Asterisk.

As these new Digium IP phones are "designed exclusively for Asterisk and Switchvox," they should remove that pain and make it much simpler to get an Asterisk solution up and running. (Side note: Does this "designed exclusively" phrase mean they won't work with other systems? Or just that they work better with Asterisk? UPDATE: Digium's Kevin Fleming answered in the comments - the phones are SIP phones that will work with any system for basic features.)

Still, the IP phone space is incredibly crowded. One vendor of VoIP products,, lists 382 results for IP phones. A quick scan of that list will show you names like Polycom, Snom, Grandstream and Aastra, all of whom have been typical phones used with Asterisk-based systems. (As well as Cisco, Avaya and other more "traditional" telecom players.)

What will these new direct-from-Digium IP phones do to the relationships with those other IP phone vendors?

Much of Digium's early business was with PSTN gateway cards that you could install into your computer. With much of that market moving entirely over to SIP trunking or SIP-based gateways, is the IP phone line designed primarily to replace that fading revenue line? Or to simply provide another revenue source for the company - perhaps at the expense of partners?

And what is the state of the market for IP phones, anyway? Analyst firm Frost and Sullivan says the market for SIP phones will continue growing and NoJitter's Eric Krapf has reported that IP phone vendors are seeing strong growth.

Still, with the "consumerization of IT" and the "bring-your-own-device" movement as people want to use their iPhones, Android phones, iPads, tablets, etc., it seems a curious move to launch a brand new line of IP phones.

However, Digium - and Asterisk - hasn't gotten to where it is by following the conventional wisdom. If anyone can carry off the launch of a new IP phone line, they may be able to do it. It will certainly be interesting to see where this takes them.

A new IP phone line... in 2012?

I would never have thought I'd be writing about that.

What do you think? Crazy move? or smart?

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Skype Issues Official Statement About The End Of Skype For Asterisk

SkypeforasteriskBefore writing my story yesterday about Skype killing off Skype For Asterisk, I had reached out to Skype's PR agency to see if there was any statement from Skype. There wasn't at the time, but today they sent over this statement from Jennifer Caukin, a spokeswoman for Skype:
Skype made the decision to retire Skype for Asterisk several months ago, as we have prioritized our focus around implementing the IETF SIP standard in our Skype Connect solution. SIP enjoys the broadest support of any of the available signaling alternatives by business communications equipment vendors, including Digium. By supporting SIP in favor of alternatives, we maximize our resources and continue to reinforce our commitment to delivering Skype on key platforms where we can meet the broadest customer demand.

Being a huge advocate of open standards, I of course applaud Skype's commitment to supporting SIP. However, as I noted two years ago in my detailed review of what was then "Skype For SIP" (and is now "Skype Connect") the fundamental difference between Skype For Asterisk and Skype's SIP offering is this:

Skype For Asterisk is/was two-way - you can make outbound calls TO Skype users.

You can't do that with Skype Connect. You can receive calls from Skype users. You can receive calls to PSTN numbers that come in across the Skype network. You can make outbound calls to PSTN numbers via the Skype network. But you can't make outbound calls to Skype users.

Skype For Asterisk could.

And therein lay much of its power.

Additionally, Skype For Asterisk passed along your Skype presence which could be used for call routing... and also supported Skype chat, too.

Neither of which Skype Connect can do right now.

Skype For Asterisk provided a 2-way, multichannel connection into the Skype cloud in a way that Skype's SIP-based offering simply doesn't at this point in time. (Having said that, of course, SFA is certainly no where near as easy to set up or understand, a point Dave Michels made today.)

However, as Alec Saunders pointed out today, the economics also clearly favor Skype Connect in terms of monthly and per-minute billing versus the low one-time fee of Skype For Asterisk. Tim Panton also indicated that the Skype For Asterisk program had some challenges including the licensing of the product.

While perhaps understandable as a business decision, I know that Skype For Asterisk will be missed by many in the technical community.

Now, let's see what Skype will truly do with their SIP support in the time ahead...

P.S. And while it is of course easy to try to blame someone like Microsoft for this demise, as I noted in my first post, the acquisition deal isn't even remotely done yet...

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Skype Kills Off "Skype For Asterisk" - A Sign of the New Microsoft Era?

UPDATE: Skype has issued an official statement about the end of Skype For Asterisk.
SkypeforasteriskWord breaking out right now from multiple sources is that Skype has killed off the Skype for Asterisk product developed in conjunction with Digium. In an email sent by Digium product management that was subsequently posted on web sites (including Digium's), the company says (my emphasis added):
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.

Skype will apparently continue to support the SFA software for an additional two years until July 26, 2013.

The Promise...


Skype For Asterisk was announced with great fanfare back at Astricon in 2008. I wrote about how it might tear down some of the walls of Skype's proprietary walled garden and posted multiple follow-up posts, including a detailed dive into Asterisk interconnection and how Skype could help with that.

The beautiful part was that Skype For Asterisk allowed two-way communication into the Skype cloud... allowing you to make calls to Skype recipients in ways that you couldn't with other options.

There was certainly great hope within the open source sides of the VOIP world that Skype For Asterisk, a.k.a. "SFA", would go far to connect the world of Skype to the larger world of SIP and IP communications.

In September 2009, Skype announced on their blog that Skype For Asterisk was available to all and there were ongoing posts on other sites about SFA usage. (Including Tim Panton's cool integration of Google Wave, Skype and Asterisk)

Sign of the Microsoft Era?

Now obviously we're not privy to the contract negotiations between Digium and Skype. Perhaps it is simply a case of the two companies not agreeing to terms. Maybe Skype wanted more money... maybe Skype didn't want to do the support for SFA... maybe it didn't hit Skype's revenue targets... maybe it's just cleaning up Skype's various business units before the Microsoft acquisition...

... or maybe it is a sign of the new Microsoft era at Skype, even though the deal has not formally closed. That is certainly the prevailing sentiment on Twitter right now.

Let's hope not... but time will tell.

Fred Posner perhaps stated this concern best in his blog post this afternoon:

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

UPDATE - May 25, 2011: Tim Panton, a developer who was among the early users of Skype For Asterisk and has been involved in the Asterisk and VoIP community for years, wrote a thoughtful post: The long slow death of Skype for Asterisk. Tim notes the apparent tension between Skype and Digium from the early days of the product and offers the opinion that Skype probably just had no intention to renew the agreement in any event. Tim's post is well worth a read as he is someone who actually worked with the SFA product a great bit.

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Calling All Asterisk Users! Can you help proofread Asterisk:The Definitive Guide?

asteriskdefinitiveguide.jpgDo you use Asterisk as a PBX? Are you an administrator of an Asterisk system? Do you have a product based on Asterisk? Or that connects to Asterisk?

If so, the authors of the forthcoming book "Asterisk: The Definitive Guide" are looking for your help as they enter into the final production stage of the book. Now, the cool part about the book is that, like the first two versions, it will be released under the Creative Commons Attribution-Noncommercial-No Derivative Works 3.0 United States license and made available online for free usage and download. You also can naturally purchase it from O'Reilly... but the key item is that the content of the book will become part of the available body of online Asterisk documentation.

So it's in all of our interest that it is as accurate as possible!

If you have even just a few minutes to browse a section or two and provide feedback, the book is up in O'Reilly's "Open Feedback Publishing System" at:

You need to have a account on O'Reilly's system in order to comment... but those accounts are free and if you have ever bought anything from O'Reilly odds are that you already have one.

In today's VUC call, authors Leif Madsen and Russell Bryant asked for help from the community. They are at the stage where they can't really add large blocks of content or massively rearrange, but they can tweak text. So they are asking people for help in just checking it over... are there any errors found? Are there better ways to say something? Text that isn't quite right? Or any other comments...


Leif and Russell stressed today that you don't need to read the whole book... if there is a chapter that interests you or that is applicable to something you work directly with, please take a look at that chapter and provide feedback. Even if you just have 10 or 15 minutes now and then to scan through some of the text, it would be a great help.

I'm going to try to read a bit of it (predictably the security chapter ;-) and would encourage you to take a peek, too!


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Tim Panton: Contrasting Digium and Skype's Developer Programs and Outreach


Longtime telephony developer Tim Panton wrote a great story this morning contrasting recent developer outreach from Digium with that of Skype:

The way to create a new product or service when you already have one.

I do agree with Tim that Digium did a great job in reaching out to the community in developing the Asterisk SCF... and I do unfortunately agree with Tim that this continues to be an area where Skype struggles. Skype is now on perhaps its 5th or 6th iteration of a "developer program"... maybe more... I've lost track, quite honestly, over all these years... still working on finding a program that builds a strong ecosystem of developers around Skype. They've hired some great people at Skype... and I'm hopeful that their newer work with SkypeKit will be positive... but we'll have to see.

[In full disclosure, my employer Voxeo has been involved with Skype's developer programs for a long time, dating back to the first "Voice Services" program back in 2005/2006 (which was later discontinued) and continues to be involved in Skype programs. However, I've not been directly involved in those programs on Voxeo's behalf.]

Tim also pointed to this great TechCrunch guest post back on November 8th about Symbian:

Guest post: Symbian OS – one of the most successful failures in tech history

The final paragraph - and final sentence - is so incredibly critical in this space:

The lesson for Meego, and other pretenders to the crown is, perhaps to look after your developers with useful APIs and powerful tools both inside and outside of your organisation. Find the right balance between efficiency and ease of development. Look after all of your developers and your developers will look after you.

Indeed... "Look after all of your developers and your developers will look after you."

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Video: AstriCon Keynote Demonstrating Asterisk Scalable Communications Framework (SCF)

Want to learn more about the Asterisk Scalable Communications Framework (SCF)? While I wrote about Asterisk SCF last week, Digium has now posted the video of the keynote session. It starts with Digium CEO Danny Windham and then at about the 5:45 mark Kevin Fleming takes the stage. At about 10 minutes Kevin starts bringing some community members on stage to tell some stories... all building up to the actual SCF announcement about 33 minutes in :-)

Regardless of the long buildup, it's worth watching if you want to understand where Asterisk is going... the demo is pretty cool, too!

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Asterisk SCF: Scalability, Extensibility, Performance

asteriskscf.jpgThe big news coming out of Astricon last week in DC was the "Asterisk Scalable Communications Framework", a.k.a. "Asterisk SCF". The main goals of the project are to bring to the Asterisk platform:

  • Scalability (and high availability)
  • Extensibility
  • Performance

As a long-time fan of Asterisk (and user/administrator at various points of time), I can agree that all of these are areas where the base Asterisk IP-PBX can use some help.

Asterisk SCF is NOT a replacement for Asterisk. Instead it is essentially a framework for extending Asterisk and adding new functionality. As the executive summary outlines:

Asterisk SCF is designed as a distributed system of components that can be deployed in clusters on a single system or on many systems, transparently. Implementing Asterisk SCF as a cluster of small components allows it to naturally take advantage of the ever-wider multi-core CPUs being produced today as well as the movement to off-site or cloud-based computing. In addition, all operational data elements required by Asterisk SCF’s components are themselves managed by their own Asterisk SCF components, allowing for active/passive failover models with no disruption of service. The design also ensures active/active failover and load-sharing models can be supported. These design elements allow capacity to be added to an operating Asterisk SCF cluster by simply enabling additional component instances.

The "distributed system" part is the key. I wasn't at AstriCon, but from the Twitter stream as the event was unfolding it sounds like the Digium folks had a good bit of fun with the keynote announcement... including where they simulated three data centers and then pulled the power from one to show that calls stayed up. This kind of capability is what Asterisk needs to grow into new areas of deployment.

Asterisk SCF is still in development... only an "alpha" is available now and a beta will be out sometime in 2011. The Digium folks have put together a pretty comprehensive wiki about Asterisk SCF at:

I found the Introduction and FAQ both quite useful to read.

As a huge believer in the power of distributed systems, I wish the folks at Digium and within the Asterisk community all the best as they undertake this Asterisk SCF effort. I (and many others, I'm sure) will be watching and looking forward to seeing what evolves out of the effort. I'm sure there will be many more posts to be written...

P.S. FYI, Dave Michels published a good post over at the UC Strategies blog: "The Next Big Thing: Asterisk SCF

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Calling all Ruby telephony developers -> AdhearsionConf 2010 starts this weekend in SF!

adhearsionconf.jpgJust one day left until AdhearsionConf 2010 opens in San Francisco tomorrow. The schedule looks outstanding and I know that the Voxeo Labs team is already on site getting everything ready. I've seen via Twitter that some hardcore Ruby telephony developers are already enroute.... should be a great time!

Sadly, my schedule doesn't allow me to be there, but the good news is that if you are unable to get there in person, you can follow along on the UStream channel:

If you are in the SF area, or can get there, check out more info about the conference at:

Great to see an event like this happening!

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Calling all Ruby and Asterisk developers! First Adhearsion Conference in SF Aug 14-15

adhearsionconf.jpgFor all of you out there working with the Adhearsion open source telephony framework to easily create communications apps on top of Asterisk using the Ruby programming language.... the first "AdhearsionConf" will be held August 14-15 in San Francisco.

Jay Phillips, the creator of Adhearsion, will be in the event and undoubtedly a great amount of hacking on Adhearsion will occur throughout the time.

The exact location and schedule are still being confirmed, but mark the date!

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