Disruptive Telephony

Nortel's fascinating move into open source telephony... but NOT with Asterisk

Nortel and "open source telephony"? Huh?

That was admittedly my thought when I received the list of who was going to be on the panel I moderated last week at VoiceCon on open source telephony. The other two panelists were obvious choices: Bill Miller was from Digium (makers of Asterisk) and Raza was from 3Com who have recently announced that they would be reselling a version of Digium's Asterisk Business Edition. Both Bill and Raza made sense to me. But Tony Pereira of Nortel? Nortel does not leap out at me as a company working with open source telephony - what in the world are they doing with it, I wondered?

It turns out that the answer is... "quite a bit!"

As Tony Pereira outlined in our panel as well as in conversations afterwards, Nortel is in the process of launching their "Software Communications Server 500" (SCS 500) targeted at small businesses and built using open source telephony software!

Interestingly, though, it does NOT use Asterisk.

Instead Nortel is using the "other" major player in open source telephony, the "sipXecs" product from SIPfoundry.org. (Previously called "sipX" but renamed "sipXecs" about a year ago.) I've not written all that much about sipX here but it certainly has been a product I've known of over the years. It started out as a PBX product from Pingtel which they then released as an open source version ("sipX" and now "sipXecs") and also had a commercial version called "SIPxchange". sipX garnered perhaps its most attention back in October 2006 when it was announced that Amazon.com would be using it for their internal phone systems (see the links on the SIPFoundry.org site). At a fundamental level, sipX provides similar functionality to Asterisk but where Asterisk is focused on being a "platform" for telephony that can work with a wide range of protocols, sipX is focused exclusively on SIP and also provides an extensive GUI management tool. (Pingtel provides a (obviously biased) comparison of sipXecs vs Asterisk on their wiki.)

From what I learned at our panel, Nortel is essentially creating their own supported version of "sipXecs" that they will sell as the "SCS 500". It will have full commercial support from Nortel. Target market will apparently be "small" businesses. No info really available on Nortel's site yet, although glimpses are visible through support documents (such as here and here(although this appears to be about an earlier 1.0 version last year)).

All in all it's to me a fascinating move by the folks at Nortel and I look forward to learning more about the SCS500 product over the next weeks and months as they launch it. It's a rather nice boost for the whole world of "open source telephony", too, to have Nortel making this move as well.

Technorati Tags: nortel, open source, asterisk, sipx, sipxecs, pingtel, voip, sip

My presentations at VoiceCon this week...

I'm down in Orlando this week for VoiceCon Orlando and will be part of three sessions. Tomorrow, I'm moderating a panel at 8am on VoIP security and on Thursday I'm moderating a panel on open source telephony. On Wednesday, I'll be part of a keynote panel with Irwin Lazar on "Social networking and enterprise communication", which should be quite fun. I'll include below the full descriptions of the various sessions. If you are attending VoiceCon and want to connect, please do contact me.

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Session Title: Top VOIP Security Threats
Date: 3/18/2008
Time: 8:00 AM
Room: Osceola B
Session Description: There's been a lot of concern about voice over IP security, but have there been many actual exploits? This session will inform you about the state of VOIP security. You'll learn about generalized IP attacks that have affected IP telephony systems deployed on IP networks, and you'll also find out what VOIP-specific attacks have actually been observed "in the wild"--and what to expect in the future.
KEY QUESTIONS: * What are the most serious voice-oriented attacks that are actually being carried out? What potential attacks haven't occurred yet but probably will before long? * How do you protect your VOIP systems against these attacks? * What types of equipment and technologies must you implement to stop voice-oriented attacks? * What specific kinds of damage can these attacks cause?
Moderator(s): Dan York - Dir of Emerging Comm Tech - Voxeo
Panelist(s): Sachin Joglekar - Vulnerability Research Lead - Sipera Systems
David Endler - Director of Security Research - TippingPoint
Mark Collier - CTO - SecureLogix

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Session Title: Open Source for Enterprise Voice: How Much, How Soon?
Date: 3/20/2008
Time: 11:45 AM
Room: Sun C
Session Description: Open source PBXs are gaining a higher profile: Asterisk and other open-source PBX software packages continue to gain acceptance, and some traditional PBX vendors have implemented open source code for their products. But these efforts still aim mainly at smaller implementations. In this session, you'll learn why open source PBX software has growing appeal, and whether it will appeal to larger customers as the market progresses.
KEY QUESTIONS: * What level of market share and acceptance has open source PBX software attained? What is expected? * Which products use open source PBX software? * What are the most compelling reasons for choosing open source PBX software? What are the greatest areas of concern in making this choice? * What are the technical challenges of an open-source PBX deployment, and how are these overcome? * What are some real-world customer experiences with open source PBX software?
Moderator(s): Dan York - Dir of Emerging Comm Tech - Voxeo
Speaker(s): M Raza - Product Management - 3Com
Bill Miller - VP, Prod Mgt & Mktg - Digium
Tony Pereira - Business Leader Business Communications - Nortel

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Session Title: Social Networking Meets Enterprise Communications
Date: 3/19/2008
Time: 10:30 AM
Room: Osceola C
Session Description: It?s no secret that world of enterprise communications is undergoing a transformation; IP Telephony and Unified Communications are changing the nature of the game. Now new forms of interaction, which began in the consumer/personal communications market -- blogs, wikis and online services like Facebook ? are migrating into the enterprise. Where do these social networking systems ? and mindset ? fit into the enterprise communications landscape? Join us for a discussion about what?s real today and what?s likely to happen in the future.
Panelist(s):
Dan York - Dir of Emerging Comm Tech - Voxeo
Irwin Lazar - Principal Analyst & Program Director, Collaboration & Convergence - Nemertes Research

Technorati Tags: voicecon, voiceconorlando, asterisk, opensource, socialnetworking, communications, danyork, irwinlazar

I'll be down at VoiceCon Orlando in March 2008...

FYI, I will be down at VoiceCon Orlando on March 17-20, 2008. I'm moderating two panel sessions (see the schedule). First, up, bright and early at 8am on Tuesday, March 18th, I'll be moderating a panel on "Top VoIP Security Threats". This should be a fun one as it has VOIPSA Chair Dave Endler, Mark Collier of SecureLogix and Sachin Joglekar of Sipera Systems. I know all three of the guys, particularly Dave and Mark who have both worked on VOIPSA matters, and this session should be a good bit of fun. I'm planning on making it a rather interactive session. :-)

At the other end of the show, on Thursday, March 20th, at 11:45am, I'll be moderating a panel "Open Source for Enterprise Voice: How Much, How Soon?". This would should be interesting because it has Bill Miller from Digium (makers of Asterisk), who I know well, and M Raza from 3Com... and then Tony Pereira from Nortel! 3Com's presence on the panel isn't particularly surprising given their relationship with Digium, but it will be interesting to see Nortel's view on the matter.

All in all it should be quite an interesting show. Lots of good sessions and, I'm sure, interesting people to meet. If you're going to be down there, please do drop an email as I'm always interesting in meeting readers of the blog.

Technorati Tags: voicecon, voxeo, voipsa, securelogix, sipera, 3com, digium, asterisk, voip security, open source

TAUG - "Asterisk and Open Telephony Conference" - Toronto, April 7-9

Those of you interested in Asterisk and open source telephony may be interested in the "Asterisk & Open Telephony Conference" in Toronto, Ontario, Canada, on April 7-9, 2008. Part of the larger IT360 conference, this Asterisk conference is being organized by members of the Toronto Asterisk User Group which is one of the largest Asterisk user groups in the world. Because of the TAUG influence, I expect it to be a solid show from a technical point of view with lots of practical information. I'm not currently planning to be there, but it looks to be a good one if you can get there.

Technorati Tags: asterisk, conferences, open source, voip, ip telephony, taug, toronto, it360

Digium launches the "Digium Asterisk Marketplace" to promote members of the Asterisk ecosystem

Digium recently announced that they have launched the "Digium Asterisk Marketplace" as a way to help connect users of Asterisk to partners in the "Asterisk ecosystem" who make products that work with Asterisk. Many of the folks listed there have been parters with Digium for some time and are often in Digium booths at various trade conferences but there were a few names new to me. At the time I am writing this, there are 30 members listed in the Marketplace but with the application form readily available, we'll see how much this grows. The Asterisk blog entry lays out the terms:

The price is right – a listing starts at $395 per quarter. In return, your company gets exposure to the thousands of unique visitors that cross the Digium site daily. Or you can get a more prominent “premium” listing for a few extra bucks. For a limited time, you can get a listing for a full year starting at $795 - about half the regular rate.

Later, we plan to add more cool features, such as the ability for users to provide feedback, more categories for listings, and the ability to buy selected partner products directly from the Marketplace site.

The question will really be how many Asterisk partners view this as a value and jump in.

In any event, I think it's great to see this type of listing coming out for Asterisk and I'll look forward to seeing how it grows over time.

Technorati Tags: asterisk, digium, opensource

Asterisk crosses 1 million downloads in 2007

News out of Digium yesterday morning that there have now been over one million downloads of Asterisk in 2007:

Digium®, Inc., the Asterisk® Company, today announced the one millionth download of Asterisk in 2007, capping off a record year for the leading open source telephony company. Digium, which will complete its 24th consecutive quarter of growth and profitability this year, created headlines with new executive appointments, industry awards, strategic partnerships and acquisitions aimed at further advancing the company's presence in the small-to-medium-sized (SMB) VoIP market.

Now, granted, we have to remember that this is one million downloads and has no real indication of: a) how many of those downloads were actually installed; b) how many of those installations were simply tests and are not in actual production use; c) how many of those downloads were duplicate downloads by the same user; and d) how many of those downloads were updates as people installed new versions.

All that aside, it is a significant milestone for any project. Congrats to the Digium team!

P.S. I'm still waiting for an answer from Digium PR about how many downloads there are total for Asterisk. This is one million in 2007, but how does that compare to the overall total?

Technorati Tags: asterisk, digium, opensource, voip

It's about the platform - Google finally answers the "Gphone" speculation... with an Android!

"It's about an open platform, stupid!" While I didn't include Google when I first wrote my post about how voice is really all about application platforms, I did note in the comments that I had intended to do so... and today's announcement really shows that they should be in anyone's list of telephony application platforms. As announced on the Google blog with "Where's my Gphone?", Google today announced the Open Handset Alliance and the associated set of forthcoming software called Android. The front page of the Open Handset Alliance provides a rather compelling (to me) statement:

What would it take to build a better mobile phone?

A commitment to openness, a shared vision for the future, and concrete plans to make the vision a reality.

Welcome to the Open Handset Alliance™, a group of more than 30 technology and mobile companies who have come together to accelerate innovation in mobile and offer consumers a richer, less expensive, and better mobile experience. Together we have developed Android™, the first complete, open, and free mobile platform.

We are committed to commercially deploy handsets and services using the Android Platform in the second half of 2008. An early look at the Android Software Development Kit (SDK) will be available on November 12th.

The list of partners in the Open Handset Alliance is quite interesting... handset manufacturers, semiconductor companies, software companies... but also cellular/mobile operators such as Sprint and T-Mobile as well as NTT DoCoMo and the giant China Telecom.

Also intriguing to note that eBay is listed as a partner. Would this be for eBay itself or could it perhaps be for Skype? Adding Skype into this mix could be interesting as well.

The Google blog page contains this text which gives some insight into Google's interest:

Android is the first truly open and comprehensive platform for mobile devices. It includes an operating system, user-interface and applications -- all of the software to run a mobile phone, but without the proprietary obstacles that have hindered mobile innovation. We have developed Android in cooperation with the Open Handset Alliance, which consists of more than 30 technology and mobile leaders including Motorola, Qualcomm, HTC and T-Mobile. Through deep partnerships with carriers, device manufacturers, developers, and others, we hope to enable an open ecosystem for the mobile world by creating a standard, open mobile software platform. We think the result will ultimately be a better and faster pace for innovation that will give mobile customers unforeseen applications and capabilities.

We see Android as an important part of our strategy of furthering Google's goal of providing access to information to users wherever they are. We recognize that many among the multitude of mobile users around the world do not and may never have an Android-based phone. Our goals must be independent of device or even platform. For this reason, Android will complement, but not replace, our longstanding mobile strategy of developing useful and compelling mobile services and driving adoption of these products through partnerships with handset manufacturers and mobile operators around the world.

In the end, Google wants a platform upon which they can offer their many services. With this plan, they are hoping to turn a zillion mobile phones into a platform which Google - and many others - can use.

Fascinating move... and one that is naturally getting a ton of coverage in the blogosphere. I've not had the time to read much of it, but did catch Scoble, the NY Times article and ZDNet's take. I'm sure we'll all be reading much more about it in the days ahead.

Right now, all we can really do is speculate until next Monday when the (apparently open source) SDK becomes available. We shall see... although the initial signs are certainly that this could indeed cause some disruption.

P.S. One of the commenters on Scoble's blog wondered why there wasn't equivalent attention being given to http://www.openmoko.com/ , especially since it is out already with a product. My quick reaction would be that from what I know of OpenMoko, it is about an open platform, but from a single vendor and on a single hardware platform. This Google announcement would appear to transcend both the vendors and the hardware platforms. It's also an announcement from Google and it has impressive backers.

Technorati Tags: android, apis, applications, google, openhandsetalliance, wireless

The audacity of Asterisk - why the 3Com/Digium partnership fundamentally changes the game in SMB telephony

The SMB VoIP game is changing. Fundamentally. And in a pattern we've seen before in other industries. In the news release out today, Digium and 3Com announced that:

Under the terms of the agreement, 3Com will offer Digium’s award-winning Asterisk Appliance™ to small businesses that need a reliable, easy-to-deploy voice solution based on open standards. 3Com Asterisk will be available through the company’s proven channel of partners worldwide.

Let's think about that for a minute. 3Com will make Digium's Asterisk appliance available through "the company's proven channel of partner's worldwide", which some reports are putting at around 60,000 resellers. Digium just wound up with a large global sales channel. Yet to be seen is whether there will be any channel conflict with existing Digium Partners/VARs, but regardless, Digium just wound up with a way to deploy Asterisk-based solutions globally. It does, however, get one step better (my emphasis added):

“3Com is focused on delivering products and solutions for converged secure networks, in which voice is an application that can be readily integrated with many others,” said Bob Dechant, senior vice president and general manager for 3Com Corporation. “We’ve announced a complete voice strategy and new product offerings for small businesses, including the 3Com Asterisk Appliance. We also offer innovative enterprise-caliber 3Com Global Services for customers who purchase the 3Com Asterisk. We chose to partner with Digium because of the company’s position as the Asterisk leader, its commitment to open standards and the ease-of-use of the appliance.”

Yes, indeed, Digium winds up with a global support organization behind Asterisk. Powerful announcement. Global sales and support - for an open source PBX... According to information from Digium, the "3Com Asterisk", priced at $1,595, will include a 3Com-co-branded interface and easy configuration/provisioning of 3Com SIP phones (as can be done today with Polycom phones). Given last weeks' announcement of the SwitchVox acquisition, I would think that rolling some of that GUI/functionality into the offering would be another logical step longer-term. The implications of this announcement, though, go far beyond the commercial relationship between Digium and 3Com. Those of us who remember Linux in the late 1990s and early 2000s remember that Linux took a trajectory like this:

1. Techies/geeks/early-adopters started to install Linux into their businesses to solve specific needs. Often it was installed without corporate permission as a DNS server, web, server, etc.
2. A range of small, specialized vendors started to ship servers with Linux pre-installed. Very often these companies were founded by people within the Linux community (ex. VA Linux, Penguin Computing)
3. Larger, more mainstream but still lower-tier manufacturers started to ship servers with Linux. (I forget the first one I saw doing this.)
4. Tier 1 manufacturers (ex. IBM, HP, Dell) started to ship servers with Linux.

Asterisk just moved to step #3 (after already moving through #1 and #2). While 3Com does not have the same market status as Cisco, Avaya and Nortel (or Mitel in SMB), 3Com definitely has a presence out there and to me their endorsement of Asterisk certainly brings a level of credibility to Asterisk-based software and hardware. It's good for Asterisk. It's good for Asterisk-based products and services (including those of Digium's competitors). It's good for open source. Ultimately, in my opinion, it's good for all of us.

Yet to be seen is how good it is for 3Com's own SMB offerings and that will be interesting to see. Right now it seems that they are all about "offering customers choice" between 3Com's own product and the Asterisk-based appliance. Will that last? Will 3Com continue to maintain its own SMB products long-term? Or will it cede that lower-end market to Asterisk and focus on apps that interoperate with Asterisk and/or phones for Asterisk (and 3Com's higher-end offerings)? Interesting questions to consider, particularly in light of 3Com's announcement on Friday that it is being acquired by Bain Capital and Chinese giant Huawei as well as their announcement today of new VCX systems targeted at the SMB market.

Nor is it clear to me how much of a short-term impact there will be on the SMB market. 3Com has been less of a presence in that space in recent years although its clear from their various announcements today that they are intent on playing a larger role in the space. Will Mitel, Avaya, Cisco, etc. lose any sales today as a result of 3Com selling Asterisk? Maybe. Maybe not.

Longer-term, though, I personally view this as a huge validation that open source telephony has a role in the business space. The cracks in the wall of proprietary telephony just got a whole lot larger today. Congrats to Digium and 3Com... and now the question is - who's the next vendor to get on board?

What do you think? Is this a validation of Asterisk? Or a flash in the pan? (Or as one more cynical person put it to me, "a desperate move by 3Com to stay relevant?") What do you see as the short-term and long-term impacts to the SMB market? Should the existing vendors be scared? Or just ignore it?

More coverage:

• Alec Saunders: Digium Scores with 3Com
• Greg Galtizine: 3Com to Offer Customers Asterisk Appliance
• VoIP-News: Digium Targets Mass Market Again With 3Com Deal
• ZD Net (Russell Shaw): 3Com fills SMB nices with new IP Telephony products
• eWeek Channel Insider: 3Com Preps VOIP Switches for SMBs

Technorati Tags: opensource, asterisk, digium, voip, 3Com, telephony

Digium buys SwitchVox and gets presence, Web 2.0 interface, mashups to Google Maps, Salesforce.com, SugarCRM...

Imagine you are a customer service rep (CSR) at a small/medium company and a phone call comes in from a customer. As your phone rings, up on your screen pops all the information about that customer, pulled from your CRM database in Salesforce.com or SugarCRM, plus other information from other databases and finally a nice Google Map showing you where that customer is located and potentially other information like the locations of your nearest offices. During the call, the CSR needs to bring in a subject matter expert so the CSR consults their web panel and looks at the presence information displayed for each of the other people in the business. The CSR can then contact someone showing as available and potentially bring them into the call.

Now imagine that all that is running on top of open source telephony... specifically Asterisk.

You can now stop imagining, because Digium just bought the company that does precisely that. There will undoubtedly be much attention today (at the very least in the VoIP blogosphere) about Digium's announcement here at AstriCon today that they have acquired SwitchVox. I am going to bet that much of the reporting today will focus on angles like these:

• Digium now has very competitive offerings (SwitchVox SOHO and SwitchVox SMB) for going after the small / medium business market.
• Digium bought themselves a very sophisticated/simple/easy GUI/management interface that moves them forward dramatically in making Asterisk easy to use, deploy and manage.
• Digium just got 1400 paying customers with over 65,000 endpoints.
• Digium bought themselves parity (or more) in their ongoing competitive feud with the folks at Fonality/Trixbox.

All of that is true. The SwitchVox products offer a very seriously competitive list of features (you have to go through and expand the subsections to see all the features). The GUI is very well done and simple. The price is quite compelling for the servers and also the support. I mean, for $1200 ($995 server plus $199 support) an SMB gets an IP-PBX with a very broad range of features and an unlimited number of users! Yes, the business still has to pay for IP phones, but they can buy any of a wide range of phones at varying price points to suit their needs. Considering that almost all the mainstream IP-PBX vendors charge on a per-user basis for licenses, the unlimited user model is certainly disruptive in its own right. (Digium has also been doing this with their Asterisk Business Edition.) And yes, Digium now has an answer to the growing competitive threat of Trixbox and it's management interfaces, support, hybrid model, etc.

All that is true - but it's not the really interesting story.

To me, what is far more compelling is that Digium just bought themselves a whole group of people who "get" the world of "unified communications", business process integration, Web 2.0 mashups, etc.

Digium has had no story at all around "presence" within its core offerings. Now it does. While Asterisk has always been a platform play where you have the ability to integrate Asterisk with other apps, doing so has not exactly been for the faint-of-heart. Hire yourself some programmers and you can do pretty much anything with Asterisk... but that's not something that many businesses want to get into. SwitchVox now gives Digium a way to do easy integration with databases and web sites. The integrations to Salesforce.com and SugarCRM are slick. The Google Maps popup is a seriously cool mashup! (And where is that on the roadmap of the mainstream vendors?)

Throw in a "click to call" add-in for Firefox to let you dial any number you see on any web page, plus a plug-in for Outlook, and you've got a very compelling offering. For a very nice price. My only knock (other than the fact that I can't find a picture of their Google Maps mashup anywhere on their website) is that it doesn't seem like their presence capability is yet integrated with existing instant messaging services. Given Asterisk's XMPP (Jabber) capabilities, this seems an obvious path that could get them connected to Jabber and GoogleTalk presence information. If they don't have that yet, I hope they add it soon, as we really do NOT need yet another place to change/update our presence info.

Regardless, this integration capability is, to me, the real story. Phones are being commoditized. I have to believe call servers/IP-PBXs are on their way to being commoditized. (Folks like Microsoft are going to help in pushing those prices down.) The money will ultimately go away from those areas.

The future of "unified communications" is about platforms. About mashups. About web services. About exposing APIs. About making it easy to combine different sources of data into interfaces that make people more productive. Microsoft gets that. Some of the traditional IP-PBX vendors get that. Digium has always known that, but this acquisition gives them a far better ability to make it happen.

Congrats to the folks at both Digium and SwitchVox for making this happen... I very much look forward to seeing where it evolves! (And in the meantime, I'm going to have to go down to the AstriCon exhibit hall and get some video of the Google Maps mashup to show how very cool it is...)

Read more:

• coverage in the blogosphere
• Digium FAQ about the acquisition
• Digium blog post by Bill Miller (more posts will apparently follow today on their blog)

Technorati Tags: asterisk, astricon, digium, mashups, opensource, presence, switchvox, telephony, voip

Zoiper - a free SIP / IAX softphone for Windows, Linux or Mac

In watching Jay Phillips do his great presentation here today at AstriCon about Ruby and his Adhearsion package, I found myself wondering what the interesting little softphone was that he was using. It turned out to be "Zoiper", an IAX or SIP softphone that was previously called "Idefisk". (I can understand perhaps why they changed the name... "Idefisk" does not exactly roll off your tongue.) There turn out to be two versions (comparison chart here): a free version and a "Zoiper Biz" version which includes more functionality and starts around 30 euros.

Clearly built for Asterisk, it was interesting to note that it supports both SIP and also Asterisk's own IAX protocol. Anyway, I just thought I'd share that this softphone is out there if you were not aware of it.

Technorati Tags: asterisk, iax, softphone, video, zoiper

Astricon... way too much to write about...

Great sessions out here at AstriCon... way too much to write about - my head is exploding a good bit! It's been a bit frustrating to try to blog from the conference rooms, though. The hotel WiFi is struggling a good bit under the strain of all the people here with laptops.

Technorati Tags: asterisk, astricon

AstriCon kicks off with "Developer 101" and "Asterisk 101" pre-conference sessions

Here in Phoenix, Arizona, AstriCon has kicked off with two pre-conference sessions that run all day. "Asterisk 101" is running next door and providing a basic introduction to Asterisk. I'm sitting in the "Developer 101" session (pictured) where there are about 100 people gathered in the room. It turns out that this is about developing with the Asterisk code base, i.e. "how to become an Asterisk developer" versus what I was personally thinking it was, which was "how to develop apps that work with Asterisk"... although that is really just an extension of the first. So far, an hour into the session, lead developer Kevin Fleming has been discussing the various tools you need to use in order to work with the Asterisk code base (ex. subversion, makefiles, etc.). Right now he's been dealing with the fun subject of licensing code, the GPL, and the requirement of developers to sign a disclaimer over to Digium that: a) asserts that the developer can contribute the code (i.e. it is original), that it is not patent-encumbered, etc. and b) gives Digium the right to redistribute the developer's code under a different license.

One interesting note - Kevin stated very definitively that Digium has NO plans to move GPLv3. They are quite happy with GPLv2 and see no reason yet to move.

The afternoon session sounds interesting as they will be getting into the overall Asterisk architecture, diving into the code, talking about APIs and debugging. Certainly a day to feed my inner developer...

Technorati Tags: asterisk, astricon, conferences, opensource, voip

Digium launches their "Inside the Asterisk" group weblog

Earlier this month, the folks at Digium rolled out their own group weblog, "Inside the Asterisk", which gives a view of activities inside Digium. A true group weblog, there are posts from a number of different people, including Digium CEO Danny Windham (ex. talking about his first days) and many others on staff.

Technorati Tags: asterisk, digium, opensource, blogs

Tom Keating loves Asterisk...

Being a long-time fan of Tom Keating's great "VoIP and Gadget Blog", I had to smile when my day started off this morning scanning feeds and encountering Tom's post on Friday "Top 10 Reasons Why I love Asterisk".  Tom first takes us (by way of a link) back down memory lane a bit with a piece he wrote in November 2001 called "In Search of a Linux-based PBX" that outlines many of the folks involved then with "open source telephony" and makes for interesting reading (his picture also looks a wee bit younger! ;-).  His piece on Friday revisits that list a bit and talks about Asterisk and how it has risen to be clearly the leader of open source telephony solutions.

It also clearly shows Tom's passion, and that is to me one of the fun and wonderful things about this thing we call the blogosphere.... writers do show more of their passion and we very often do get to learn more about the people behind the names.... and that makes it a whole lot more interesting than just dry and blase "factual" articles.

So kudos to Tom for telling us why he loves Asterisk...

Technorati tags: tom keating, asterisk, open source

TMC.net launches "Open Source PBX community"

Recognizing the growth in open source telephony, Rich Tehrani and his team over at TMC.net today rolled out their "Open Source PBX community" at http://opensourcepbx.tmcnet.com/ . Sponsored by long-time open source supporter Sangoma, Rich says in his video that they aim for their site to be the place to go for news, commentary and documents about open source telephony.  Right now it seems primarily to have some white papers, a list of resources (which seem to mostly come from Sangoma) and open source-related news and articles.   Given that this just launched, it will be interesting to see what they do with it.

Rich asked for feedback about the site and my #1 point would be - could we please get some RSS feeds?  Unless I missed them, I can't see anyway to subscribe... and that's really how I personally like to read my updates these days.

Anyway, it's good to see TMC.net expanding their open source coverage and I'll be watching how it evolves.

Technorati tags: voip, open source, asterisk, sangoma, rich tehrani, tmc.net

Digium/AsteriskWorld to be co-resident with VON Fall 2007 in Boston

In my queue of things to write about, I'd be meaning to comment on the fact that this year's Fall VON on October 29 - November 1 in Boston will also be the site of Digium / Asterisk World.   Jeff Pulver's team announced this back in March and it will have Mark Spencer and others from Digium involved.  I wasn't particularly surprised to hear this, given that Jeff Pulver has always had an interest in open source and that Digium has always had a large presence at VON exhibit halls.  (You would have to be blind to miss the giant orange booth with its many pods!)  Kudos to Digium and Pulvermedia for organizing a way to spotlight all the great stuff happening in open source telephony.  I'll look forward to seeing what goes on there.

Technorati tags: asterisk, digium, asterisk world, jeff pulver, von, von2007

New site promoting the integration of the Ruby language and telephony...

By way of a Twitter post today, I learned that Dave Troy has unveiled "Talking Ruby", a new site promoting information about the integration of the Ruby language with telephony, collaboration and messaging.  I've always been intrigued by Ruby (and also Ruby on Rails, which has been one of the most visible uses of Ruby), but have yet to really have had a reason to plunge in and play with it.  Perhaps this will provide an excuse.  Dave indicates on his site the following reasons for using Ruby with telephony:

• Ruby’s DSL (Domain Specific Language) Capabilities are ideal for expressively encapsulating diverse telephony and collaboration technologies
• Inherits the momentum of Rails, so web integration is baked-in
• Cross-platform support (Linux, OS X, BSD, Windows ) unifies application development efforts
• Ruby integration libraries can be easily developed and shared
• DRb (Distributed Ruby) allows for persistent state storage and scaling across servers

I wish him all the best with the new site and do look forward to seeing what people come up with.  The site is a wiki, so if you're interested and Ruby-literate, you can easily jump in and participate (there's also a mailing list).

Technorati tags: voip, ruby, telephony

Asterisk running on a Roomba - "Press one to start sucking"

Here's something for a Friday afternoon... yes, indeed, out at Emerging Telephony 2007 back in February, someone (David Troy) did indeed have Asterisk running on a Roomba.  And yes, it was "Press one to start sucking. Press two to stop sucking."

Pictures are now available on Flickr.

More precisely, Asterisk is running on a hacked Linksys WRT54G access point (which is Linux-based) and the controller is using a Nokia WiFi/GSM phone to connect to the Asterisk install.  You could also control the direction of the Roomba using the other keys on the phone keypad.

Why would anyone do this?  Well... why not?

As I mentioned in a recent post, the beautiful thing about VoIP is that it now enables people to "play" with telephony... and do wacky things like hook it up to a Roomba!  :-)

Enjoy the weekend!  Perhaps next week I'll actually get some time to upload the rest of the pictures I took out at ETel. (Hey, it was only 6 weeks ago... )

Technorati tags: asterisk, roomba, voip, play, open source, telephony

VoIP Now confirms that the cool kids hack telephony with their list of 74 open source VoIP / IP telephony projects

As I travel around giving presentations about the technologies that are disrupting telephony, one of the themes I discuss is that one of the most severe disruptions brought about by VoIP is that people now have the ability to "play" with telephony in ways that were never possible before.  Pre-VoIP, you needed special (and typically costly) equipment.  Yes, there have been any range of CTI cards that let you play to a degree, but buying the real equipment was just not possible for most folks who might want to "hack" in the original meaning of the word.  Enter VoIP.  Now all you need is an old PC and some open source software and... ta da... you're playing with telephony. 

What I also see out there is that this ability to hack on telephony is happening at the same time that hacking on networks or operating systems seems to be getting less exciting and interesting.  Oh, don't get me wrong, there's still amazing things happening out there... but for people who want to "play" with technology, those areas aren't as exciting or novel as they once were.  So many of those early adopters have moved on to hack on other things... primarily, it seems to me, on "Web 2.0" apps/services/mashups... or telephony.  (And you'll note the already happening collision of both.) 

Because I like giving presentations with very minimalist slides (unless forced to bulletize), I often summarize this latter point as:

The cool kids now hack telephony.

Whether you agree or disagree with my point, I don't think anyone can deny the continued growth in the number and capability of open source telephony projects.  By way of voiploser's blog (also worth checking out), I learned of VoIP Now's list of 74 Open Source VoIP Apps and Resources.  It's a great list, which really serves to illustrate the amount of open source activity happening with regard to telephony.   Some of the projects on there have been around for quite a long time, while there were certainly some there that were quite new (and I'd not heard of them).

My only quibbles with the article would be these:

• There appears to be no way to leave comments, which is too bad, because you would undoubtedly get all sorts of other developers coming out of the woodwork and leaving comments saying "Hey, what about my project?"
• No matter how you structure a list, people will always say it's wrong.  So naturally, I question why you would start with "H.323 Clients" given that all the major work these days is on SIP.
• Given my past interaction with FreeSWITCH (read the comment left by the lead developer), I somehow doubt that they would want to only be classified under "H.323 Clients".  In fact, the inclusion there really makes absolutely no sense to me given that FreeSWITCH is decidedly not a client, but rather more a telephony platform.  It should probably have gone down near the PBXs or in a separate "platforms" category.  Ditto with YATE.  Part of me wonders if the author just wanted to list FreeSWITCH as #1...
• Under "SIP Test Tools", they list some of the more prominent ones, but the VOIPSA "VoIP Security Tools" list has far more, most all of which are also open source and are used to "test" your VoIP system.

Quibbles aside, the list is definitely a good one, and kudos to VoIP Now for putting it together.

Technorati tags: open source, asterisk, voip, ip telephony, voice over IP, voice, telephony, telecommunications

ETel: FreeSWITCH Boot Camp...

I admit to really only very peripherally followed the growth of FreeSWITCH, so I was intrigued to attend the "FreeSWITCH Boot Camp" session this morning here at ETel.  It was a tough call given that Stowe Boyd was also speaking, but I wanted to understand what FreeSwitch was all about.  It was an interesting talk, although I'm left with the following observations:

• I'm still struggling to fully understand what problem the FreeSwitch community is trying to specifically solve versus what Asterisk, sipX, OpenSER, etc. are solving. 
• The answer from the FreeSwitch developers was that it is really complementary to those other projects and focus on scalability and stability.  It is NOT focused on the PBX space but really at the carrier space and looking at large-scale implementations.  Several people also mentioned using it as a Session Border Controller (SBC).
• So is it an open source SBC?
• One carrier representative involved with the project indicated that in their testing they are getting 2,000 to 3,000 simultaneous calls up with media streaming... and at least 10,000 simultaneous calls with point-to-point media.
• Perhaps that is the focus... but I would say that the FreeSwitch folks need to refine that message bit so that it's a bit easier to understand.
• Management is still pretty much all through config files.  Web GUI is still "in the works".
• Looks to have a pretty comprehensive list of protocols, codecs, application interfaces, etc.
• What was perhaps most interesting was their web-based interface to a conferencing system.  Pretty nicely done.

Overall, my impression was that it's an interesting toolkit to let folks play with telephony on potentially on a large scale.  It will be quite interesting to see what evolves out of the FreeSwitch developer community.  I'd be interested to know if anyone reading this is using FreeSwitch and what they are doing with.

Technorati tags: voip, freeswitch, open source, asterisk, etel2007, etel, emerging telephony

Anyone out there using ChanSkype to connect Asterisk to Skype?

Anyone reading this blog using the ChanSkype software to connect Asterisk to Skype?  I've not played with it at all myself, but it sounds like an interesting idea.  Here's what they say it can do:

• Call online Skype users.
• Call using SkypeOut.
• Receive up to 30 incoming Skype Calls ("Skype Trunk").
• Bridge with SIP channels.
• Make any number of simultaneous calls (limited only by system resources).

Their FAQ is just a wee bit sparse on details, like, oh, precisely how many simultaneous connections will it support?  Their main page has the text above and on the Buy page they note that corporate licenses are licensed per port up to 30 users and it has this text:

This limitation is not technical, for ChanSkype's simultaneous call capabilities are limited only by system resources.

Which naturally makes me a bit more curious.  It's clear that they are using the Skype client-side API through a Linux Skype client but that's about it.  I would think to support multiple users they would have to launch multiple instances of the Linux Skype client.  Is this what they are doing?

If anyone has played with it, I'd be curious to know how it works.  It's intriguing enough to me that I might just have to revive my dormant Asterisk install.

Technorati tags: asterisk, skype, voip, chanskype

Heading to Ottawa... and OCLUG's Asterisk talk tonight

In just a few minutes I'll be getting in today's rental car and heading up to Ottawa for the remainder of the work week for some meetings at the corporate office.

One of the nice things about being up there today is that I'll get to drop in on an OCLUG meeting tonight.  In the most of five years we lived in Ottawa, OCLUG was a wonderful place to meet some really incredible people.  I very much enjoyed the time spent there and the friendships that were formed... and I look forward to seeing many of those folks again tonight.   Not quite sure where the "Beer SIG" will be since the meeting is now out on Woodroffe, but I'm sure there's an appropriate spot somewhere around.

The meeting will be doubly interesting because tonight's topic will be Asterisk, which of course is of great interest to me.  Should be interesting to see what is being discussed. (Not quite sure what Randal will be doing with "Fractal Poetry", but hey, it sounds intriguing, anyway.)

If any of you reading are going, I'll see you at the meeting.

(As I now get in my white rental car to drive up to Ottawa with snow all around... whoever thought up creating white cars obviously missed the fact that in the winter those cars are almost invisible.  Ugh.)

Technorati tags: ottawa, oclug, asterisk, linux

Mark Spencer changes roles at Digium/Asterisk... new CEO

Wow!  Some big changes down in Alabama... per the news release yesterday (hat tip to Alec Saunders), Mark Spencer is changing roles within Digium and stepping aside as CEO to bring in ADTRAN's Danny Windham as the new CEO.  Mark will remain as Chairman and CTO of Digium.  They also announced a new worldwide VP of Sales.  Tom Keating has a good writeup of the announcement and also has a link to a podcast of the conference call announcing the changes.  It makes for good listening.

As noted, ADTRAN has had a long relationship with Digium... if I recall correctly, Mark started out as an intern there at ADTRAN and when he went on to launch Linux Support Services and needed a phone system, it was his background at ADTRAN that made him think that if he could just get telephony onto a PC, he could manipulate it.  Thus was born Asterisk.  (And then later Digium, the company, to support Asterisk.)

I'm glad for Mark to see the change.  I've come to know him a bit over the years through conferences/trade shows and the interviews we've done with him for Blue Box...  and he's definitely a hard-core techie and developer... he's done the CEO role because it's needed to be done, but his talents really are in the technology and I'm sure this role change will let him get more back into the technical stuff that is his passion.  Congrats, Mark!  You've certainly deserved it and I do wish you all the best.

Technorati tags: digium, asterisk, mark spencer

Friday afternoon video #1: Mark Spencer introduces AsteriskNow

It's a Friday afternoon... and so I thought I'd end the week with two videos I've been meaning to write about.  First up, Mark Spencer shows off their new AsteriskNow distribution:

So if you've never seen Mark Spencer, there he is.  AsteriskNow seems to be Digium's answer to Trixbox, now that Fonality purchased Trixbox and has been very actively promoting it.  (Trixbox, you may recall, was originally called "Asterisk@home".)  The idea is similar... package Asterisk and all other necessary components with a Linux distribution and provide a very simple GUI with wizards to install the whole thing.  Looks simple from the video... I'll have to try it out once I get vmware or something else running on one of my systems again.

Technorati tags: asterisk, digium, mark spencer, voip, open source, asterisknow

O'Reilly asking community to help create "Asterisk Cookbook"

I love it when I see the collision of a variety of my interests.  Here O'Reilly combines VoIP, open source and social media (a wiki in this case) all in one effort: Bruce Stewart is looking for people to "Help Create the Asterisk Cookbook".  Here's his request:

We’re looking for two kinds of contributions. First, we’re looking for problems you’d like to see solved in the book. If you need to make Asterisk do something and just can’t figure out how, let us know. We’ll try to solve the problem for you. Second, we’re looking for more advanced Asterisk users to contribute solutions to problems that they’ve faced.

And Bruce has a wiki set up for people to use to contribute.  This is interesting on a couple of different levels.  First, it's a major publisher going out to a community to directly involve them in writing a book. Second, they are using a wiki for all the collaboration. (And yes, other authors have reached out to communities and have used wikis for public collaboration, so this isn't necessarily a new idea.)  And third, the topic being Asterisk, I'm sure they'll wind up getting recipes from a ton of people who have scratched their various itches and solved peculiar problems using Asterisk.  In fact, it will probably could have been titled "The Itch-Scratcher's Guide to VoIP."

I'll be intrigued to see the result, not only of how the collaborative process works for O'Reilly, but also for the actual book.  The fascinating and fun part about Asterisk is that because the code is wide open for anyone to tweak to their heart's content, people can scratch itches and solve problems that are so particular to them that no commercial vendor in their right mind would ever spend the time or resources to address the issue. There's just no real market for it beyond that one company/organization. But that entity can turn to Asterisk and either program it themselves or pay someone to develop the feature or fix for them.  If they make their code public, it might just turn out that there are some others out there who might have the same or a similar itch.  And the itch-scratching continues...

So it will be fun to see what recipes emerge.

Technorati tags: asterisk, voip, bruce stewart, oreilly, emerging telephony