Druid: Open Source Unified Communications Based on Asterisk

A group called Voiceroute just pushed out a product called Druid (actually two products, an open source version and a commercial version) that looks very intriguing. It is based on Asterisk at the core but they've done a lot of work on top of it -- besides just sticking on a useful GUI. This is what I love about the telephony space today -- continuous improvement as folks figure out what they really want. There's a decent little preview/review here. A more thorough explanation of features here. Some ideas for applications here.

So many forms of communications (voice, voicemail, email, IM, mobile, fax), so many ways to leverage it. Now a single IP communications platform based on open source and open standards.

What is Druid? Druid is the premier unified communications platform for enterprises. It allows companies to deploy easily and affordable high endIP communications services using off the shelve commodity hardware and IP phones. Druid covers your enterprise communication needs from IP voice, voicemail, IM all the way to the mobile space.

I love Asterisk and the various things which can be be built with it. But, at its core, (stock) Asterisk excels most at being an engine at the center of a telephony/communications platform. To maximize adoption, the barrier to use must be lowered through improved management interfaces and turnkey features out of the box that reflect what end-users want. There's lots of room for improvement for specific problem domains. That's why Asterisk's flexibility is so powerful and having a lively eco-system around it makes it more and more accessible to a broader base of users. That doesn't mean that Asterisk based solutions aren't ready for prime time today -- they most definitely are. Thankfully, Asterisk's eco-system is strong with solutions such as Fonality, FreePBX, Digium's AsteriskNOW/ABE/AA50, and many others. Oh, and now we have Voiceroute's Druid.

Definitely will be taking a look Druid soon.

Improvements at Digium? ....and New AA50 Firmware

Yesterday I noted that Digium pushed out a new firmware release for their small business targeted hardware appliance (the AA50) and it looks like a good sized update. It is dated the 4th in the release notes but the 14th elsewhere so I think one or the other is a typo.

They appear to be syncing it up with their other code bases (Asterisk Business Edition, AsteriskNOW). From what I've seen in the public commits and hints on mailing lists from staff, they've been hitting pretty hard on both of those in the QA department of late.

Digium is clearly working on changing how they manage their code bases and the mixture of their commercial and open source releases, to address different customer/user segments and some of the problems that have come up. There was even a note about a split release change they are going to with AsteriskNOW for the GPL purists. Glad to see some serious re-thinking about releases, meeting the desires of different customers, quality control enhancements, and the public evidence of the aforementioned outside of press releases.

I imagine things will continue to trickle down and we'll see more improvements coming together. I'm sure some will turn out to be good ideas while others head back to the drawing board but that's how most progress works, eh? :)

The additional leverage should improve things across the board no matter where you fall in the Asterisk eco-system. Even if you don't use an Asterisk based telephony solution, their moves impact the others in the marketplace (and visa-versa).

Back to the new AA50 firmware release: It would appear to address IVR and ring group related bugs I encountering early on in the GUI, that I ended up bypassing the GUI to workaround. I haven't tried it yet on the AA50 I have access to. I'll probably give it another couple o' weeks in the wild first since it's a .0.1 rev. That client has some other problems and improvements they want me to attack first anyhow. Below are change log excerpts:

Version 1.1.0.1 - February 4, 2008
* Enable Internationalization settings in the GUI
* Provide the ability to select between kewlstart and loopstart
* g722 codec is available
* WAN Side Provisioning of Polycom phones is enabled
* AA50 has been synchronized to ABE C.1 branch source code
* Ring groups number of seconds field has been added
* Ring group bug using ivr option has been fixed.
* Polycom bug concering use of standard timezone has been fixed
* Blackfin math for meetme conferencing has been implemented
* Adjustable flash hook duration is now available in teh GUI
* Calling rule editing and deleting bug has been fixed.
* Bug concerning setting incoming rule to choose voicemail is fixed.
* DTMF twist settings for Brazil have been added to tone generation
* Call forwarding loops have been prevented so that they do not crash the AA50
* Voicemail attachments are set to WAV format.
* Network setting tabs no longer disappear.
* Added an optional full-wave mode DAA ring detect in the sx00i driver

Asterisk Mashes Up Politics

I ran across this application today, called CommitteeCaller.com, which makes it easier for (U.S.) folks to contact their representatives. It's a nifty example of the type of applications that become possible when some imagination gets combined with lowered barriers to entry. This is what mashups are all about. Taking information that is out there on the Internet and combining it in ways that make it more useful, accessible, relevant, visible, etc.

This particular one uses Asterisk for the telephony, a database built from information on the Internet, and a custom AGI to interact with the user input, look up things in the database, make the calls, and get post-call rating feedback. AGIs are the equivalent of HTTP world CGIs (yes, the Asterisk world is progressing quite fast but the Web did get a big head start on it so it's still a little behind; CGIs, or AGIs, are pretty 1997 but you have to start somewhere).

Just wait until all the old school web developers that are used to coding in PHP, Ruby (Adhearsion), C, Perl (Asterisk::AGI), etc. discover they can write Asterisk telephony applications just as easily and in the same languages. (The Adhearsion page, even if you're not a Ruby programmer, has a good overview and example applications if you're curious).

CommitteeCaller.com is a site that allows one person to target an entire congressional committee over the phone. The web application utilizes the open source Asterisk PBX system to connect you to every senator or house member on a particular committee. No more digging around the 'net entering zip-codes to retrieve phone numbers of representatives. CommitteeCaller.com automates the tedium of finding and dialing your favorite politicians.

Select a committee, enter in your phone number and click "Put me in touch with democracy!" and you'll be called by our system and sequentially patched through to the front office of each member on that committee. You can even rate how each call went; information that will enable us to rank representatives on how accountable and responsive they are to their constituents.
[...]
Once connected Committee Caller will tell you which representive you are calling, who their legislative director or chief of staff is, and what district they represent. At any point you can use the * to hang up the call and move on to the next one. Remember not to hang up after each call as you will have the opportunity to rate how your call went.

-jr

Cordless VOIP Phones, Well, Stink

Michael Graves, over at the VOIP Users Conference, has a nice summary of the common frustrations he's had with cordless VOIP (SIP) phones. Michael plays with a lot of hardware, is a real user, and isn't selling anything related. His opinions are worth a read.

-jr

My New Home Asterisk PBX Embedded Box

[ The first two are photos of the actual unit, pulled from the eBay auction. The last two photos are not mine but stolen from a friendly Flickr source. Hoping Santa will bring the family a new camera.. ;-) ]

On Friday my HP Thin Client arrived. Only I'm not going to use it as a thin client. Instead I'm going to install Linux (or FreeBSD, see below) along with Asterisk, the open source telephony platform, onto it. This particular unit is an HP T5700.

It cost me <$100 on eBay. Low power <20w, class="blsp-spelling-error" id="SPELLING_ERROR_1">Ghz (Transmeta Crusoe) based with 256MB RAM, and 256MB Flash. I may add a 2.5 laptop drive (has IDE and USB too).

The lack of noise and the tiny form factor is a huge driver since this is going into our apartment and the low power is for our savings accounts and the environment. :) The flexibility and ease of use gained by keeping it x86 based is a big plus for saving time (no cross compiling, no searching out funky code patches for less mainstream architectures) and maintaining compatibility with the maximum amount of things I may want to do with it.

I nearly bought an IP04, or one the variations based on it like uCpbx. These are Blackfin based embedded systems designed to run Asterisk, specifically the Astin distro. These are very cost effective looking solutions for SMB type environments. They are also very very similar to Digium's Asterisk Appliance 50 (AA50), which is also Blackfin based. In fact, they are nearly the same thing if you don't count formal Digium backing and support. (I recently got some experience in with an AA50, in an installation for a client with Snom 320 phones and intend to post some about that at a future date).

I'm looking forward to getting this box on-line as our full-time home phone system. If all goes well I'll probably pick up another one (or several) for lab use. This should free up some space in the apartment and not require me to keep shutting my dev box down to eliminate ambient noise and power consumption. Now if only the power supply would get here quick. :)

Typically I install Debian or Ubuntu then plop in Asterisk. This time, this is meant to be more of a true "appliance" than a server. So I'm going to evaluate some other options before I settle on anything. This will give me the opportunity to get experience with and thus cross out some items on my "To Evaluate/Learn" list.

I'm going to try out Askozia which looks promising. It'll be a new one for me and it's actually FreeBSD, rather then Linux based (it's based on m0n0wall). Askozia also has a built-in web GUI which I'm looking forward to contrasting with Digium's own GUI (which is in AsteriskNOW). AstLinux is another option I'll check out. Unfortunately out-of-the-box it's Asterisk 1.2.x not 1.4.x based (though there's a dev version that is 1.4.x).

I do miss having a lot of tools anytime I've worked with embedded distros. And I like having extensive logging available -- even if the device is supposedly an "appliance" that just sits there. Tough to troubleshoot an appliance when there ain't no logs. :-)

Ultimately I may roll my own stripped down something or other. Or, grab a more generic already stripped down distro and put Asterisk plus the Digium Asterisk GUI on top. Having the option to add a laptop drive gives me comfort I can go with this route, even as far as installing Debian or Ubuntu stock if need be, while remaining low-power.

I also intend to experiment with the various Bluetooth integration options for Asterisk. Namely chan_mobile for headset and cell phone integration.

We'll see what else. :) The nice thing is that, besides really enjoying Asterisk, I can justify more than one solution, since whatever I don't actually use as the house system will still be rewarding in the lab for self-education and evaluating the options out there for my clients.

-jr

The Modem Free Generation

Wow, how time does fly. We now have, at least in the US and elsewhere, our first generation of Internet users that have no idea what a real modem sounds like. For these folks, this is the closest they'll ever get to one.

While I'm not as old as the photo above by any means, my first modem was an external U.S. Robotics 2400 bps (generously loaned to me by a friend's father whom I later bought a more affordable Gould 1200 bps modem from), so I suppose, in a way, even I was late to the game.

Sure, with Caller ID we still have modems in our phones (at least until end-to-end SIP can do away with all that) and xDSL is still really a glorified analog modem but they are stealthy. Poll a random nine year old on the street with a modem carrier audio sample or ask if they've ever cursed when they forgot and set ATM2 instead of ATM0 or ATM1 and you'll get a blank look (and probably a scream for mommy to come and take them away from the scary crazy guy though a few smart top-of-their-class nine year olds, just starting their introduction classes to CCIE certification of course, might think I'm talking about this which is at least in the right vein).

I've yet to see an xDSL modem that has a speaker (let alone supports the AT command set) and rarely do I hear the caller ID carrier unless I'm on a really cheap phone and pick up the phone fast enough.

Maybe we can bring it back in vogue by customizing our mobile phone ring tone to sound like the good old days. And, demanding that xDSL modem vendors, add speakers. On other hand, there aren't too many of these around anymore either. And, man, were those things slow.

AT
OK
ATZ
OK
ATM0
OK

-jr

Improving the Snom IP Phone Retrieve Button Functionality

(Otherwise entitled "Getting the Snom Retrieve Button To Work Even When There Are Only Old Messages to Retrieve")

If you use Snom IP phones, you may have discovered, as I recently did, that the Retrieve button doesn't work, at least by default, unless there are new messages in the mailbox. If the user wants to listen to a saved voicemail, they are out of luck and have to dial the special voicemail extension directly. The Retrieve button just sits there and does nada.

This created some confusion in a recent installation since folks learned to use the Retrieve button to access voicemail and then later, after a day or so of use, wanted to listen to voicemails they had saved. :-) Sure, I had set-up an extension to dial the voicemail system directly (intended for when people were out of the office) but it was pretty silly to have two different ways for users to get used to accessing their voicemail, depending only on whether they had new messages or not.

I poked around a bit and the fix was very simple.... In the web management interface go to Setup->Identity X-->Mailbox and set it to your internal mailbox extension.

(As an aside: I am overall pretty happy with the Snom 320 IP phones. Be careful what firmware revisions you are running -- stay away from 7.x unless you know what you are doing and keep things consistent across your installation).

VOIP Troubleshooting With (the free) Wireshark Packet Analyzer

Wireshark is a network protocol analyzer. Some may recognize it by its former name, Ethereal. It's free (and open source), runs on multiple platforms (including Windows and Linux), and actively developed. For those doing VOIP installations or troubleshooting existing installations, the latest release has some very handy VOIP specific support.

It will create visuals representing captured SIP and associated RTP connections. You can drill down by clicking on specific spots on the graph to pull up the associated packet(s). You can generate reports (as well as graph) jitter, bandwidth usage, etc. Various ways of displaying the data to get a better idea of what's really going on.

The screen captures at the beginning of this post are from Wireshark. They show a graph of a VOIP (SIP) call (and a half) between two Snom SIP phones attached to an Asterisk-based PBX (the green/blue/purple image). And an analysis of the associated RTP session (including packet loss, jitter, delay). WS can even playback captured VOIP calls (at least if using PCM/G.711/ulaw).

-jr

Digium (Asterisk) Is Sending Busy Signals

Digium, the commercial company behind the open-source Asterisk IP PBX, has been ultra busy of late. They came out with a self-contained hardware based Asterisk appliance targeted at developers, telephony carriers, etc. to build custom IP PBXes for their customer bases. They followed this with a full blown out-of-the-box installable hardware IP PBX appliance (the AA50) intended for the mass market. They bought SwitchVox, a leading IP PBX appliance vendor with some nice innovative user interface and functionality features, then announced a deal with 3Com who will be OEM'ing their appliance as the foundation of their IP PBX offering.

Digium, has been selling components, such as as cards to interface between the traditional phone network, development and support services for Asterisk, and commercial licenses for vendors OEM'ing Asterisk code into their own PBXes and other telephony applications for several years now. Now, it's time to get serious I guess.

Further Notable Links:

• http://www.disruptivetelephony.com/2007/09/digium-buys-swi.html
• http://www.voip-news.com/feature/digium-3com-deal-100107/
• http://blogs.digium.com/
• http://seekingalpha.com/article/48678-3com-corporation-forget-cisco-think-open-source
• http://www.digium.com/en/mediacenter/news/
• http://www.digium.com/en/mediacenter/news/viewpress.php?id=3com-and-digium-partner-to-deliver-3com-asterisk