The VoIP Experience

Time to get busy with VoIPology stuff.

2011-10-18T15:31:00.000-07:00

Wow... Have I let this blog go...

Well I do have a pile of new gear to review.

So, watch for reviews of the new Aastra 6737i, Peplink B20 router and an op ed piece on why I think the cooperation between the FreePBX Project and schmozecom.com is a good thing.

Where did FreePBX v3 go ?

2010-08-03T04:55:00.000-07:00

The other day I say a post in one of the VoIP forums asking "What happened to FreePBX v3?" and I got to wondering that myself. Well it turns out the project is alive and well... It just has a new home. I caught up with Darren Schreiber, the lead developer on the project and posed a few questions. Here they are with Darren's responses:

Q: There seems to be a void of information about FreePBX V3 status.
Is 2600hz a continuation of the v3 project or is it something completely different or both ?

First of all, 2600hz is an overall foundation and project. It will be the home to many projects in the future.

blue.box, a project that is now a part of 2600hz, is a rework of the v3 project. A good amount of code has been replaced and the core and modules have been made more stable. We are going to further overhaul the framework, as we've learned a lot in the past few years working on the code.

Q: Previously we saw that FreePBX V3 was going to support FreeSWITCH,
Asterisk and possibly others. Does the new direction include supporting other engines besides FreeSWITCH?

Yes, we will support both engines, but our focus will remain on where the demand has been. So far, most people who are happy with Asterisk and FreePBX v2 have been less interested in FreePBX v3 and FreeSWITCH. If that trend continues, so will our focus.

Q: 2600hz is born from TCAPI, would you care to elaborate on the history of TCAPI and how it is now 2600hz?

The original TCAPI site is still up, and while I only finished a few pages, the code behind those pages was complete. It involved the concepts and functionality around the MVC architecture, splitting of modules/plug-ins, including a rich JavaScript layer with helpers and so-on. The blue.box project still uses the exact same XML and configuration generation code that was originally introduced in the TCAPI framework.

Q: 2600hz is a new company, what are the short and long term goals ? What "space" do you anticipate operating in ?

Short term, we hope to build some killer GUIs and VoIP configuration and management tools. Same as always. Long-term, we hope to make VoIP easier. We really mean that. VoIP is too complex for what it is and for the expectation people have for solid, reliable voice communications services. We intend to change that.

Q: Lastly, anything you want to mention wrt this new venture ?

We hope to see all projects - FreePBX v2, FreeSWITCH, Asterisk, blue.box and others - continue to flourish. I think there's room for many, many projects that will fill many different needs. Our objective is VoIP that is reliable and scales, and that's our main goal going forward. Make VoIP Simple.

Ufda, I need to write more consistently....

2010-07-23T15:39:00.000-07:00

I get so caught up in what you might call "other stuff" that I forget to micro publish now and again.

Time to goad myself into thinking of something interesting to write about.

Any ideas ?

Trying out Sidewiki

2009-11-28T09:10:00.001-08:00

I usually start with New Posts to see if there are any interesting developments...

Now, I am just testing Sidewiki and see it this ports in my blog ?

in reference to:

"New Posts"
- PBX in a Flash Forum - Powered by vBulletin (view on Google Sidewiki)

FreePBX v3 Ramblings....

2009-08-05T17:56:00.000-07:00

I find it interesting how things get twisted in the wind via the blogosphere.

Such as this slightly misinformed bit nestled in in a blog and I quote:

Significantly, FreePBX v3 is telephony-engine agnostic, and able to support FreeSWITCH, Asterisk and others.

Well, that is not quite right. More like FreePBX v3 "will be" telephony-engine agnostic. At the moment they are starting with FreeSWITCH and for some very solid reasons such as scalability and geek factor. Developers and Engineers like the latest technology which in turn should help gather more open source development goodness/traction. Not to mention that Asterisk is currently very well supported by FreePBX 2.5 and soon 2.6 and phone systems arguably need to be on the slow upgrade path. I see some are trying to install v3 on Asterisk based boxes already... I wonder if they read that blog?

In the installs I have done of FreePBX v3, I saw indication of future support for not only Asterisk, but YATE as well. So, my best guess is that it may hold up at FreeSWITCH, Asterisk and YATE. Not only that, but I have to think that the ability to build communications infrastructure that leverages those engines for their strengths to be a lot more powerful then the sum of their parts.

Right now FreeSWITCH essentially has nothing in the GUI arena. To be fair there is a pfSense module and possibly the Cudatel GUI from Barracuda Networks. I have not seen Cudatel. What I have seen and done is write FreeSWITCH dialplan in nano and while quaint, was not going to get me using FreeSWITCH for very much other than simple testing. I have a lot on my plate and just can't go there too much.

Anyway, another blog starts out:

With its third-generation release, out this week, popular open-source management interface FreePBX has taken a bold step away from its longstanding ties to Asterisk, to embrace the up-and-coming VoIP alternative FreeSwitch.

Is Digium sensitive to this? Of course they are. Is this a "bold step away?" Uh, Asterisk support is announced as well... The general move seems engine agnostic. I don't see the point in leading in that way. Seems misleading.

At any rate Asterisk support will be some time soon. My guess is that it may be six months before this is ready for production installs. Hopefully, it will be a harmonious time.

Another announcement from Bandwidth is the Developer Sandbox. I am probably not enough of a coder to get to play, but I am very interested in some of the other tech I am familiar with. FreePBX of course, but SMS, HD and T38 are interesting. Fixed mobile convergence is really interesting...

I'll report back if I get to dig in with my Tonka or not.

FreePBX v3 announcement

2009-08-05T06:55:00.000-07:00

FreePBX Version 3 has been announced. Listed in the announcement are some of the goals of the next gen GUI:

* Make FreePBX truly telephony engine independent
* A well thought out modular design building on the past 5 years of experience
* Use of a true MVC framework
* Choose best in breed components and build on top of them
* Excellent documentation to encourage developer involvement
* Easily Skinnable

Here is a screen capture of my first install:

I did a second run through this time and before doing the FreePBX install bit I made sure the php-soap and php-xmlrpc stuff was installed.

There are some major changes in the underlining "concepts" from FreePBX 2.X to 3.X. So, time to get engaged in the IRC, forums and the lab. Looks like ciphering time Jethro...

Stay Tuned

Link to announcement

Cisco Unified CM 6.1 to Asterisk as SIP Proxy to Bandwidth.com

2009-06-07T09:43:00.000-07:00

One of the systems I manage is an 875 Extension Cisco Unified Call Manager(UCM). At the moment the system uses SCAN trunks for long distance calling. These SCAN Trunks are provided by the state of Washington and interconnect via a four port FXO card. Callers use a PIN to make long distance calls. This is some seriously old school technology and as such has sound quality to match.

So, armed with a four port SIP trunk account from Bandwidth.com, I set forth to make a SIP trunk from the UCM to Bandwidth.com. Long story short... it does not look like I can. I can set things up that should work, but don't. There is an active TAC open with Cisco and when(if) we make it work then I'll be back in a new blog entry.

Instead, we have "Cisco Unified CM 6.1 to Asterisk as SIP Proxy for Long Distance service."

What is difficult to impossible with UCM is trivial in Asterisk w/FreePBX.

For Bandwidth.com I made a new SIP Trunk with the name of "freepbx" and here are the PEER Details:

username=myusername
type=peer
sendrpid=yes
secret=mypassword
qualify=yes
insecure=very
host=trunk.freepbx.com
fromdomain=trunk.freepbx.com
context=from-trunk

The Register name was formated like this:

myusername:mypassword@trunk.freepbx.com/360746XXXX

-------

For the UCM, I made a SIP Trunk named "ucm" and here are the PEER Details:

type=peer
context=from-callmanager
host=10.XXX.XXX.XXX
disallow=all
allow=ulaw&alaw
nat=no
canreinvite=no
qualify=yes

and in extensions_custom.conf I added ( to have these calls bypass most of the dialplan logic ):

[from-callmanager]
include => from-internal
exten => _1NXXNXXXXXX,1,Dial(${OUT_12}/${EXTEN});
exten => _1NXXNXXXXXX,n,hangup

NOTE: the the "OUT_12" reflects the "freepbx" trunk and the last thing to do is allow anonymous SIP as the calls are routed w/o authentication from the UCM ( for now.)

-------

Now on to UCM.

Since we are using no authentication to send calls from UCM to Asterisk this part is somewhat straight forward. First step is to define a "SIP Trunk Security Profile Configuration" making sure the "Outgoing Transport Type" is UDP, which looks like:

From CallManagerToSIP

Then we create the SIP Trunk, noting that the "Calling Search Space" in this case is "Reserved Incoming Calls" which basically does not allow inbound calls to go anywhere:

From CallManagerToSIP

The last three items to set up are the "Route Group, Route List and Route Pattern."

Here is the Route Group:

From CallManagerToSIP

Here is the Route List:

From CallManagerToSIP

And since we need a way to direct calls out this SIP Trunk to make long distance calls, here is the Route Pattern so users dial "891NXXXXXXXXX" to dial out:

Bare in mind, I only want calls going out this trunk ( not in) so if you wanted bi directional calls... You would adjust the "Calling Search Space" in the trunk to allow it.

Otherwise, that is the setup that is live for now. Once an Authenticated SIP Trunk can be sorted out on the UCM, I plan to go directly to Bandwidth.com and skip the proxy.

April 2009 Open Telephony Training Seminar Recap

2009-04-24T08:06:00.001-07:00

First of all I want to apologize for the formatting of the three previous posts. I am fussing around with a new editor and I am unable to wrangle it without mangling the layout. I guess I should re-label my blog as "beta."

Now that I have a chance to rest and reflect, I feel pretty good about how the Open Telephony Training Seminar went. It was my first time at the helm and I think it is safe to call the event a success. I'll be preparing a Google app survey a bit later today, so it will be interesting to see what people have to say in writing. To be sure, I received excellent advise from Philippe Lindheimer and Tony Lewis which made the task a bit easier.

The big take away for all of us(trainers/learners) is that a tranquil setting enhances training and is less fatiguing for the duration. It is a credit to the proprietors of the Willows Inn and by extension the Bed & Breakfast as a conference center.

At the end of the day, the only mishap was one person got the wrong lunch. Other than that, the lodging, catering, furnishings, electricity, network, lab equipment, phones, projector and laptops all worked without fail for the duration.

You don't get to say that every day.

Open Telephony Training Seminar Day 3

2009-04-23T14:23:00.001-07:00

Day three of the Open Telephony Training Seminar and a shot of me doing some instruction.

Of' course I have to slip in a view of the Chocolate Pate we had as dessert last night. The main course was fresh halibut and it was absolutely delicious. It would appear that our learners got quite a bit out of the instruction and doing this kind of training in this environment was of a real and tangible benefit.

One of the comments was about how quickly the days seemed to go by. I too felt that way and was a bit surprised. No matter how interesting a subject may be... At the end of the day the minutes drag on. That just didn't happen.

The breakfast the last morning of the show.... I think the quality and quantity of the meals had a bit to do with how well the training days flowed. We may be on to something here.

I would like to thanks the attendees for uh... Attending. Also, a big thanks to Judy, Riley and the whole Willows Inn staff for rolling out the red carpet for us.

Further, I want to thank the founders of the Open Telephony Training Seminar - Tony Lewis, Ethan Schroeder and Philippe Lindheimer for allowing me to organize a west coast venue.

Thanks also to Digium and Aastra for providing some very nice swag for attendees. Each attendee went home with of the just release 6730i's and a TE122 Series card. A few went home with Willows Inn Robes too.

In closing, a huge extra thanks to Philippe for co teaching the course and otherwise being a great friend.

Open Telephony Training Seminar continued...

2009-04-22T12:17:00.001-07:00

One the eve of Day One when we met for evening Cocktails, we had a nice treat waiting for us. Spot Shrimp season has just started for the Lummi Nation fishermen. Riley Starks, one of the owners at the Willows Inn, being the ever vigilant proprietor secured an amount of these "1st of the Season" treats for our enjoyment.

I forgot to get a picture of the smoked salmon we had just before the Shrimp came out.

Here you can see Riley demonstrating the method of consumption.

Since I live out here, I took a step back and let our guests dive in. I did have a couple as they wound down a bit.

In general, I'd have to say that it is working out as planned. While it was a bit farther for students to travel, the location is perfect for small classes.

We are now into Day Two and everyone seems to be absorbing the material well.

For lunches, we go downstairs and enjoy some nicely prepared meals in the Tap Root Cafe. Which, because of the quality, happens to be one of my favorite spots for lunch. Below is Philippe digging back in.

The Open Telephony Training Seminar starts today

2009-04-21T12:17:00.001-07:00

Even in this economy we managed to attract enough students to hold this months OTTS. So, we are off to a really good start. One of the bigger issues we are having is the view.....

Yes, it is a small class, but that was the intention of this "Special Venue" edition of OTTS. We have a good mix of students and they appear to be very studious. I'll have to get some photos of the meals we are being served. Here you can see Philippe Lindheimer doing his instructor bit...

Here is another view of the classroom. I want to send out a special thanks to the Willows Inn for the tremendous hospitallity they are demonstrating.

Check back as I chronicle the rest of the show...

WarVOX, friend and foe alike.

2009-03-28T10:00:00.001-07:00

Naturally, as more Voice over Internet Protocol systems and devices are placed somewhere in the vicinity of well... the Internet, more interesting things are likely to occur. A look through your firewall logs has been a semi frightening activity for some years now. In the world of Asterisk based VoIP and you had better be watching not only your firewall logs, but your Asterisk, Apache, MySQL logs and your CDR for added measure. Fortunately, we have tools such as OSSEC, fail2ban and Ethan Schroeder's excellent Abnormal Call Volume script ( see repost at end of article) to help us out.

While I routinely use Nessus to scan for vulnerabilities, I am always on the look out for new tools to test systems with and one of the new kids on the block is WarVOX. At the end of the day, WarVOX is an automated dialer, with roots to the old school autodialers of yore. Telephony hacking has been around for a long time and WarVOX is a another tool to assist you in auditing your DID's. On the other hand, you can also view WarVOX as something the dark side is going to be using (in force ) to map out things to exploit in some fashion. What do you think? "Would you like to play a game?"

While there has been some blogoshpere coverage, I hadn't seen much around building and using WarVOX, so using a nice hosted instance of Ubuntu 8.10, I used the following commands to make ready and install ( recreated the order from the log after a dependency battle):

sudo apt-get install libstdc++6
sudo apt-get install gcc-4.3-base
sudo apt-get install ruby1.8-dev
sudo apt-get install libopenssl-ruby1.8
sudo apt-get install build-essential libiaxclient-dev sox lame ruby rake rubygems libsqlite3-ruby gnuplot

gem install mongrel

Then...

wget http://warvox.org/releases/warvox-1.0.0.tar.gz
tar -xzvf warvox-1.0.0.tar.gz
cd warvox-1.0.0
make

Change the username and password in /warvox-1.0.0/etc/warvox.conf and if all goes well you can launch it with:

/bin/warvox.rb --address yourIPaddress --port 7777

Once it is running open a web browser and go to that address with port 7777 and after login you will be greated by the home page of your new WarVOX:

Now, you have to set up a few providers, at warvox.org they suggest the following:

When thinking war dialers, you usually were limited by the number of simultanious conntections to the PSTN you could make. WarVOX uses IAX providers which eliminates the need for PSTN connectivity and, if you have enough bandwitdh and multiple accounts, you can really speed up testing large blocks of DID's.

In testing you need to be mindful of the target. 555-XXXX will dial 10,000 numbers, where 555-555X will dial only 10.

The number of seconds is set at 53 seconds as a default. The concept there is that providers don't charge for calls that are not answered. In testing I found that many calls did go to voicemail and therefore constitute an answered call or more to the point - a chargeable call. Two cents each on Teliax. It is advisable to test this out and check you providers call records to find a number of seconds that works more to your favor, but still gets the job done. Of'course if you use multiple providers, this becomes a much trickier balance.

The maximum number of outgoing lines is considered to be limited by bandwitdh-->provider(s)-->trunks. I would imagine that horsepower of the box would play a part here, but that in most cases the number of trunks one can afford would bottom out before the CPU does.

The source Caller ID is probably another place to be mindful. Not all providers are going to allow spoofing a Caller ID. On the ones that do, you can use an area code and prefix that makes them standout in the CDR if you like. I use one of my own DID's as I only test systems I am responsible for. ***Note: Just as with any other "whitehat" style of system probing, prodding and cataloging - you might consider having some form of written permission to perform such tests on systems you do not own.

Once you hit the create button the job starts...
The page will refresh occationally until the job is finished and once the yellow bar goes away you can then check the results.
The next course of action is to click on the Analyze Calls which results (after several seconds) in this display.
The Analysis of Job ID 14 contains some interesting data including an audio files to listen to. At the moment, my test system is only maintaining one recording per job. If I scan a range, the only thing that plays back is the first thing recorded. The others appear to have been recorded in the log, so maybe I can sort that out. I did contact the developers and am hoping for a bug fix soon.

In larger scans, I was able to identify FAX and Voice lines and some curious time outs. Which was useful because a manual followup call to those timed out DID's showed I had a few, out of a block of 850, that were not functioning properly.

In conclusion, WarVOX is shaping up to be a very powerful tool and something that we should be on the lookout for in our CDR's. One thing I noticed is that while Caller ID is randomized, it seems to use just one ID per job. So, if you see a succession of calls through you DID's from one number... WarVOX maybe the culprit.

This topic, along with some of the other security topics mentioned will be covered during the next Open Telephony Training Seminar, April 21st~23rd in Seattle,Wa. This particular event, with lodging and gourmet meals included, is being held at a special venue called The Willows Inn and we will have the place to ourselves. This is a good event to bring a spouse. They can relax while you learn. Do consider joing us.

Contact me via email - rkeller at legoebayuc.com - for special discounts to the April 2009 Open Telephony Training Seminar.

(Reprinted of Ethan Schroeder's excellent script)
Well, since this thread got deleted and it contained really good information and was linked from my blog, I thought I would re-post it.
This is in reference to the following news story: http://www.news.com.au/technology/story/0,28348,24939188-5014239,...
As a result of this article, I wrote a script that runs once a day and sends email alerts if call volume increases in any of the following 4 areas:
1.) Total outbound calls in the last 24 hours is higher than the threshold % versus average outbound calls per week day over the last 30 days
2.) Total international outbound calls in the last 24 hours is higher than the threshold % versus average outbound international calls per week day over the last 30 days
3.) Total outbound call duration over the last 24 hours is higher than the threshold % versus average daily outbound call duration per week day over the last 30 days
4.) Total international outbound call duration over the last 24 hours is higher than the threshold % versus average daily outbound international call duration per week day over the last 30 days
To download and install:

wget http://public.schmoozecom.com/Abnorm...0-1.noarch.rpm
rpm -Uvh AbnormalCallVolume-1.0-1.noarch.rpm
service crond restart
nano /usr/local/sbin/abnormal.php

Once editing the abnormal.php file, change the email address and if you would like daily reports regardless if thresholds were met, change $daily_report = false; to $daily_report = true; If you only want to receive reports if thresholds were reached, leave this as false. You can also change the threshold percentages if you so choose. By default an email alert gets triggered if any of the four areas described above increase by 20% or more in a day

Skype for Asterisk, Google Voice, OTTS, FreePBX and Digium

2009-03-20T16:47:00.000-07:00

Gabcast! The VoIP Experience #71 - Skype for Asterisk, Google Voice, OTTS, FreePBX and Digium

Skype for Asterisk
Currently at version 0.94 and climbing. Works, while there is a lot of added functionality that Skype provides ( Chat, sms, etc...) The focus is making the telephony end perfect 1st, then address other functionality.

Google Voice
I did gave a Grand Central account and did not really find it all that useful. I did get spim on the DID... Now that I have converted GC to GV I am pleased so far. I set up some new phones for the DID to ring when called and that all seemed to work fine.I really like the Call and SMS Functionality. I can text, but I think it looks silly so I avoid it. Now, I can SMS with my normal tools. Below is a few screen shots of the Call and SMS functions.

The voice recognition of voicemail was also interesting, if not entirely accurate. It will be fun trying the transcription service out. For me it did not like my annunciation of 11 and thirty. The transition from speakerphone to handset was a bit rough too. But, I expected much worse and I expect it to improve over time.

OTTS

April 21st ~ 23rd , 2009. 1st time in the Pacific Northwest for the Open Telephony Training Seminar. Philippe and I are co teaching and the Willows Inn is a very special venue with meals and lodging included. There are a few new bonus items that come along with the major bonus of the training itself. Upon arrival you will be greeted with a choice of a) one Willows Inn Robe -or- b) a wine flight with both dinners -or- c) a one hour massage at the spa. Complements of FreePBX.org

This is the time the classes start to fill up and there are very reasonable arrangements for spouses to come along and enjoy the meals. Please do join us.

FreePBX
It feels like FreePBX 2.6 is warming up in the ball pen. One of the interesting new features in the ability to choose the module respositories. Standard or Extended. Some examples of an Extended module would be Bulk Extensions or Set Caller ID, etc... I will be reporting back on developments as they occur.

Digium
Digium really seems to be a community player to me. Because of the relationship that Tony, Ethan and Philippe have deveoloped with Digium, I have seen them participate and sponsor the Open Telephony Training Seminars, including providing training space in their Hunstville HQ. The SFA beta program is the 1st "sorta" inside experience I personally have had working with Digium and I have found the whole group to be very personal and I am having a good time with that too. Well, I haven't been to Digium HQ... yet.

Open Telephony Training in the Pacific Northwest

2009-03-02T19:56:00.001-08:00

Sometime early in 2006 I started building systems with AA@H. About the only inexpensive way to learn Asterisk was in the forums and blogs. It was in the Nervittles blog of Ward Mundy that I won entrance to the first Fonality trixbox® Open Communications Certification (FtOCC, pronounced “F-talk”) workshop, held in LA - March 2007.

The best thing that came out of FtOCC was putting faces to names I seen online. The technical level of FtOCC ended up being a bit too basic for me, and I started to ask around the greater Asterisk community if anyone, including Fonality, had plans to hold more of an enginners course in the near future. Fonality was dead silent. Philippe Lindheimer the lead developer of the FreePBX project, whom was taking over for Rob Thomas at the time... wasn't.

Philippe and I engaged in lengthy discussions and I even enticed the some of the FreePBX team out for a gourmet meal to talk about what would make the ultimate Asterisk/FreePBX seminar. As things progressed, I was introduced to Tony Lewis and Ethan Schroeder of Schmoozecom (a successful Asterisk based solutions provider) when FreePBX and Schmoozecom colaborated to create the Open Telephony Training Seminar or OTTS for short. The 1st of which February 27-29th , 2008 in Charleston, SC.

I helped shape the OTTS experience with FreePBX and Schmooze, providing my parts of the expertise and producing parts of the content that I have spoken publicly on in other venues and I was invited along to teach the E911 section of the course. It was a roaring success and a great time was had by all.

Fast forward to 2009 and I am organizing the first OTTS in the Pacific Northwest. On April 21st, 22nd and 23rd, 2009 the trainers at FreePBX.org have cooked up something very nice for the next offering of the Open Telephony Training Seminar as the venue is located in the northwest corner of Washington State , halfway between Seattle and Vancouver, BC sits Lummi Island, home of The Willows Inn , an historic Bed and Breakfast retreat. FreePBX-athon redux. This time Philippe Lindheimer and I will be teaching this course in a slightly modified version to fit the venue. Most of the marketing section has been omitted and the class is three days instead of four. It will feature a live lab for demonstration and learner use during the course.

The Open Telephony Training Seminar provides some of the best Asterisk®/FreePBX® VoIP training available and has an additional constant source of enjoyment - trainers and learners tend to stick together through meals and enjoy good conversation into the evening. These adhoc roundtables are not planned, it just happens and is said to be one of the great joys of the Open Telephony Training Seminar.

Learners are in for a treat at the Willows Inn as its organic, authentic farm-to-table fine dining is consistent with Slow Food, making The Willows Inn one of the most sought after B&Bs and agritourism destinations in the San Juan islands. And just as small class size benefits our children in school, small class size in technology training is also better for the adult learner and with chef prepared organic meals the trainers and learners will have plenty of time together. Arrangements can be made for spouses to attend meals and otherwise relax as their significant other learns.

After the seminar, On April 26th and 27th the tenth annual LinuxFest Northwest 2009 is being held in nearby Bellingham, Wa. As an added bonus, learners who want to stay and attend Linuxfest will be offered weekday rates that weekend at the Willows Inn. Book early if that is something you are interested in.

We need to fill ten seats to make our new format a reality and as these events are always great, the value is there. The Willows Inn has allowed us to block off their entire facility, but it means filling it up to make the class a reality. So with that note, we need to get 10 participants – and are offering a 20% discount for the first 3 to sign up by March 15th date to make sure we reach that goal. Contact me at rkeller at legoebayuc.com for the coupon code.

Please consider joining us.

The VoIP Experience #70 - Interview with Philippe Lindheimer - Bandwidth.com sponsors the FreePBX Project

2008-11-17T15:21:00.001-08:00

The VoIP Experience #70 - Interview with Philippe Lindheimer - Bandwidth.com sponsors the FreePBX Project

This is an interview with Philippe Lindheimer about the recent announcement regarding Bandwidth.com and their sponsorship of the FreePBX Project.

If you have additional questions for Philippe, post them at http://voipexperience.blogspot.com/

The VoIP Experience #69 - FreePBX and Bandwidth.com join forces.

2008-11-14T12:52:00.001-08:00

The VoIP Experience #69 - FreePBX and Bandwidth.com join forces.

Just a short bit about the announcement from FreePBX that they and Bandwidth.com have joined forces.

The VoIP Experience #68 - Pika WARP now with FreePBX

2008-11-07T17:25:00.001-08:00

The VoIP Experience #68 - Pika WARP now with FreePBX

The Pika WARP appliance is beta testing FreePBX for use in managing this exiting appliance.

First of all, I am not going to do the usual "Out of the Box" review. This gentleman has this covered. I am going to be reviewing usability and function. As well as some stress testing for fun and profit.

Other than the aforementioned blog, there are some interesting threads h ere and there.

Before the unit arrived, I received a post from Shawn Karow at Pika letting me know that Beta 2 is out and I should flash the unit with that load... Oh great... Each kind of appliance has a procedure for placing new loads on the box and it was time to learn the proceedure for the WARP. After reading that some of the directions out there may not work I was a bit concerned... Turns out that there is an autorun script available.

First you take an empty Fat32 formatted USB drive and place the autorun script and the files from the beta download on the root of the USB drive.

Second, as root, you ssh into the WARP and do an "amportal stop" and insert the USB drive. The update occurs automatically.

Was much easier then expected. At any rate, I connected an Analog handset and an Aastra 57i, configured them and made a few calls. Hmmm, the FXS can call the SIP phone... But the SIP can't call the FXS? I remember Philippe Lindheimer mentioning something about how ZAP is a bit different. For instance, on a current (read - non DAHDI) i386 PBX in a Flash install you would see "ZAP/2" but that needs to be changed for the Pika WARP to "PIKA/fxs/1" or "PIKA/fxs/2" etc... A small change, but if you didn't know where to look and what to change it might boggle the mind for a while.

This may get interesting as I add an analog trunk or two. I would imagine that the format is "PIKA/fxo/1" and so forth.

The only other thing I have done at the moment was add a Teliax SIP trunk, inbound and outbound routes and an IVR. Speak of which the stock FreePBX IVR module is for MySQL and does not work with SQLite3... However, there is a thread in the Pika forum that leads to a new IVR module that does appear to work.

So, I flashed, done some configuration of the Asterisk dial plan via FreePBX and made calls in and out of the system.

I'd say we are off to a good start.

I am going to have to be careful and only load the modules I need.

Next up... Its hammer time...

The VoIP Experience #67 - Continued melt down on trixbox forums

2008-10-18T17:19:00.001-07:00

The VoIP Experience #67 - Continued melt down on trixbox forums

I was tipped off to look at a thread on the trixbox forums about a botched upgrade to the pbx admin portion of their product and was alarmed by the thread of Tim the Engineer. The melt down is in full melt...

Here (in case it gets deleted) is the missive to which I speak:

What the heck. I have never piped up on the FreePBX/tbm-pbxconfig debate but this is pretty much the boiling point for me.

Andrew, he is not attacking us/you/trixbox or anyone. Damn man, we do take his code and rebrand it. We strip out anything the resembles FreePBX and then spin it. I know, I roll the RPM. The comment of "If the same issue occurred on the FreePBX fixing it would take me approximately 15 seconds" is more than likely true and shows just how inept this fork is.

I reported this issue to you and Kerry last week. When I reported the module issue yet again to you this morning your comment was less than receptive "Send me the link". I have all but had it with the attitude between trixbox and Philippe. I have heard both sides and am fed up with the lot of it.

I told you guys not to do this from the beginning, damn near begged. I tried yet again after astricon to get you guys to roll back to normal FreePBX, but that just would not happen.

Why? Because of branding? Because it says FreePBX where it should say FreePBX (Not a freaking typo)? Has anyone had a problem selling a free trixbox system because it said FreePBX in certain sections of the GUI?

Ahhhhhhhh this whole thing just pisses me off.

Thank you,
Engineer Tim

--
http://engineertim.com

And here is a link to the thread.

The VoIP Experience #66 - Philippe Lindheimer of FreePBX on AsteriskNow 1.5

2008-10-13T12:50:00.001-07:00

The VoIP Experience #66 - Philippe Lindheimer of FreePBX on AsteriskNow 1.5

This is a short interview with Philippe Lindheimer of FreePBX in which we talk about the just concluded Open Telephony Training Seminar at Digium Headquarters. Also, we talk about the new kid on the block-AsteriskNow 1.5 with FreePBX 2.5.

Also, make sure the check out the Digium Blog as it has additional info about OTTS and AsteriskNow 1.5.

The VoIP Experience #65 - Asterisk 1.6 and DAHDI

2008-10-06T08:25:00.001-07:00

The VoIP Experience #65 - Asterisk 1.6 and DAHDI

A short narrative about the Release of Asterisk 1.6 and that it is DAHDI only.

My question is does that make Asterisk 1.6 Digium hardware only for a while?

The VoIP Experience #64 - Interview with Philippe Lindheimer of FreePBX

2008-09-28T19:32:00.001-07:00

The VoIP Experience #64 - Interview with Philippe Lindheimer of FreePBX

The topics in this interview include the recently released FreePBX 2.5 and some of the enhancements that this release has brought to the utilization of Asterisk, including the announced AsteriskNow 1.5 with FreePBX. This combined with the FreePBX Open Telephony Training Seminar at Digium Headquarters next week promises to keep this fall interesting.

The VoIP Experience #63 - Summer is over and I am back.

2008-09-28T18:25:00.001-07:00

The VoIP Experience #63 - Summer is over and I am back.

Time to get back to work.

DAHDI, no jokes version with FreePBX

2008-09-07T14:51:00.000-07:00

It has been a while since I have blogged and I guess it is about time to get down and funky...Or better yet, how about down and DAHDI ? So much for the no jokes rule. At the very least, that one is original ;)

DAHDI or Digium Asterisk Hardware Device Interface is (or at least should be) a name change away from Zaptel. The reasons are found here.

All that aside, I am simply going to explain how I am testing DAHDI.

I started with a PC w/TDM400 and one FXS module and loaded PBX in a Flash 1.3 brought FreePBX up to 2.5.0rc2.4 and then hand compiled Zaptel 1.4.12, DAHDI-2.0.0-rc3+rc2 and Asterisk 1.4.22.rc3.
The interesting part is that FreePBX 2.5 ( when released ) will have the ability to use Zaptel or DAHDI. No joke. On the left is a snippet from my test box. Zaptel gone... DAHDI in.

OK, "DAHDI in" if you follow my basic tenant of IT goodness, which is "Surround yourself with really smart people." So, ahem... DAHDI in thanks to some help from Philippe Lindheimer.

Basically, the DAHDI installation introduces a few new configuration files. I am going to list the directories and the contents and then the contents of my files.

/etc/dahdi/ "init.conf , system.conf , modules"

and

/etc/asterisk/"chan_dahdi_additional.conf , chan_dahdi.conf , dahdi-channels.conf , chan_dahdi.conf.template"

So, first the /etc/dahdi/

init.conf:
#
# Shell settings for Dahdi initialization scripts.
# This replaces the old/per-platform files (/etc/sysconfig/zaptel,
# /etc/defaults/zaptel)
#

# The maximal timeout (seconds) to wait for udevd to finish generating
# device nodes after the modules have loaded and before running dahdi_cfg.
#DAHDI_DEV_TIMEOUT=40

# Override settings for xpp_fxloader
#XPP_FIRMWARE_DIR=/usr/share/dahdi
#XPP_HOTPLUG_DISABLED=yes

system.conf:
# Autogenerated by /usr/sbin/dahdi_genconf on Sun Sep 7 12:00:22 2008 -- do not hand edit
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)
# channel 1, WCTDM/4/0, no module.
# channel 2, WCTDM/4/1, no module.
# channel 3, WCTDM/4/2, no module.
# channel 4, WCTDM/4/3, no module.

# Global data

fxoks=1 # internal phones

loadzone = us
defaultzone = us

echocanceller=mg2,1,2,4

modules:
# Contains the list of modules to be loaded / unloaded by /etc/init.d/dahdi.
#
# NOTE: Please add/edit /etc/modprobe.d/dahdi or /etc/modprobe.conf if you
# would like to add any module parameters.
#
# Format of this file: list of modules, each in its own line.
# Anything after a '#' is ignore, likewise trailing and leading
# whitespaces and empty lines.

# TE405P - Quad Span T1/E1 Card (5v version)
# TE410P - Quad Span T1/E1 Card (3.3v version)
# wct4xxp

# * TE120P - Single Span T1/E1 Card
# wcte12xp

# T100P - Single Span T1 Card
# E100P - Single Span E1 Card
# wct1xxp

# TE110P - Single Span T1/E1 Card
# wcte11xp

# TDM2400P - Modular FXS/FXO interface (1-24 ports)
# TDM410P - Modular FXS/FXO interface (4 ports)
# TDM800P - Modular FXS/FXO interface (8 ports)
# wctdm24xxp

# X100P - Single port FXO interface
# X101P - Single port FXO interface
# wcfxo

# TDM400P - Modular FXS/FXO interface (1-4 ports)
wctdm

# DAHDI timing only interface. Disabled by default, as it will be loaded
# by the init.d script in case no hardware is found.
#
# If you have no hardware, enable it and disable others.
#dahdi_dummy

# Xorcom Astribank Device
# xpp_usb

And now the /etc/asterisk/

chan_dahdi.conf.template:

;# Flash Operator Panel will parse this file for dahdi trunk buttons
;# AMPLABEL will be used for the display labels on the buttons

;# %c Dahdi Channel number
;# %n Line number
;# %N Line number, but restart counter
;# Example:
;# ;AMPLABEL:Channel %c - Button %n

;# For Dahdi/* buttons use the following
;# (where x=number of buttons to dislpay)
;# ;AMPWILDCARDLABEL(x):MyLabel

[channels]
language=en

; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf

; XTDM20B Port #1,2 plugged into PSTN
;AMPLABEL:Channel %c - Button %n
context=from-pstn
signalling=fxs_ks
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
channel=1-2
dahdi-channels.conf:
; Autogenerated by /usr/sbin/dahdi_genconf on Sun Sep 7 13:02:22 2008 -- do not hand edit
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/asterisk/chan_dahdi.conf that will include the global settings
;

; Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)

chan_dahdi.conf:
;# Flash Operator Panel will parse this file for dahdi trunk buttons
;# AMPLABEL will be used for the display labels on the buttons

;# %c Dahdi Channel number
;# %n Line number
;# %N Line number, but restart counter
;# Example:
;# ;AMPLABEL:Channel %c - Button %n

;# For Dahdi/* buttons use the following
;# (where x=number of buttons to dislpay)
;# ;AMPWILDCARDLABEL(x):MyLabel

[channels]
language=en

; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf

chan_dahdi_additional.conf:
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;

;;;;;;[7000]
signalling=fxo_ks
pickupgroup=
mailbox=7000@device
immediate=no
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callgroup=
callerid=device <7000>
busydetect=no
busycount=7
accountcode=
channel=>1

Something that should be noted is that as DAHDI is really only being introduced to us, so is it just being introduced to FreePBX. So to speak. For now, we will still be speaking ZAP while we may be using DAHDI. The next picture is a GUI representation of the code in chan_dahdi_additional.conf.

But, all ( I think) I really had to do was stay out of chan_dahdi_additional.conf and let FreePBX do its job. That and instead of issue genzaptelconf... issue dahdi_genconf to get the auto configuration action going.

There are a few other commands to note:
ztcfg -> dahdi_cfg
ztmonitor -> dahdi_monitor
ztscan -> dahdi_scan
ztspeed -> dahdi_speed
zttest -> dahdi_test
zttool -> dahdi_tool

A couple of other things to note are this may not work yet with third party cards. I have only tested a Digium card and only one at that.

The FreePBX / DAHDI Integration that will come with 2.5 will of'course help speed along the adoption of DAHDI. When I first started to reseach where I was going wrong , I found only three posts on the Digium forums about DAHDI. I guess every one is smarter than me and didn't need any help...

There are quite a few more new features in FreePBX 2.5 that are of interest. This one is a good first step for DAHDI adoption rates. As always, I have more to report, but it must wait just a bit longer to break...

Open Telephony Training Seminar - Vegas

2008-05-21T08:53:00.000-07:00

Dateline Las Vegas... We are now in day 2 ( actually day 1 as we had a day 0 for new Asterisk users ) and things are moving along very smoothly. On day zero the basics of FreePBX and the Linux CLI were discussed and we helped all attendees get their VM sessions ready for the Labs. We didn't do labs during the last training , however... we are doing them this time. So far, so good.

I'll continue to take photos and keep up with the action. There are a lot of vendor interaction from companies such as Digium, Aastra, Redphone and others.

trixbox forks FreePBX

2008-05-17T06:28:00.000-07:00

Gabcast! The VoIP Experience #62 - trixbox forks FreePBX

It looks like the folks at Fonality have forked FreePBX claiming the bugs are not getting fixed, etc... What a bunch of BS!

Fly by Nights hurting our collective Asterisk reputation

2008-05-10T07:17:00.000-07:00

Gabcast! The VoIP Experience #61 - Fly by Nights hurting our collective Asterisk reputation

A short rant about FBNs and their negative impact on the VoIP industry as a whole.

LinuxFest NW 2008 post show...

2008-05-03T17:44:00.000-07:00

Previously mentioned... I volunteered to do a presentation on Open Telephony solutions and was asked to do a telephony lab for the fest. When I arrived to set up for presentation I found a nice room with a podium and the projector on and ready. I set up and soon the room was standing room only. I managed to stretch a 20 minutes of material in 90 minutes of what had to be one of my best presentations to date. I hope someone shares a photo so I can share it here.

Overall I felt good about taking the time to prepare for and give a presentation about a topic I am passionate about. As a volunteer I also felt good that the organizers did a good job of advertising and attracting people to the event.

I just wish the part I was asked to perform went as well. It could be that the distinction between "volunteer" and "requested of" is important here. At any rate, my response to the request for a small lab was room to project and place for four to six handsets....

I showed up to find that no one had any knowledge of this "telephony lab." Nice... I readied a completely portable iPBX and procured handsets and such. So, after a period of time it looked like there was some no shows for the exhibit floor and "the guy" pointed at a table and said here take that one. I started to say I needed an Ethernet drop when "the guy" interrupted me to blurt that "he knew what I needed." Mind reading with a touch of politeness... Cool...

Actually, it was a bit surreal. Ah, but Jacob Perry did manage to photograph my "bad booth."

"Too bad" is probably more like it. As an SMB lab set up it could have been a contender. As a booth... not so much. I had to get creative. It wasn't too long before someone noticed that I didn't appear very linuxy. I hit the bar and all they saw was the Asterisk CLI..

Actually, there is a side shot in the top photo.

Asterisk and the Policy Block list

2008-04-16T18:00:00.000-07:00

I got a call from a client that stated his vm to email stopped working. So, I looked about and nothing seem too wrong. Then I read the asterisk email... Seemed that Qwest had started blocking.

Now, in a way I support this as it helps to slow the tide of zombies. Think Kraken...

Here is the entry from Spamhaus:

"Ref: PBL171468

75.172.0.0/17 is listed on the Policy Block List (PBL)

Outbound Email Policy of Qwest for this IP range:

It is the policy of Qwest that email sent from a dynamically assigned IP address should be sent out only via the designated outbound mail server allocated to Qwest customers. To find the hostname of the correct mail server to use, customers should consult the original signup documentation as provided by their ISP or contact Qwest Technical Support.

Removal Procedure

If you are not using normal email software but instead are running a mail server and you are the owner of a Static IP address in the range 75.172.0.0/17 and you have a legitimate reason for operating a mail server on this IP, you can automatically remove (suppress) your static IP address from the PBL database."

So, I went thru the policy removal and in about 30 minutes email started to flow again.

It is still a time bomb as the IP seems to change a lot. So, I should probably dig out Ward's gmail solution and have a go.

Linuxfest Northwest 2008

2008-04-11T19:56:00.000-07:00

Gabcast! The VoIP Experience #59 - Linuxfest Northwest 2008

I'll be presenting and demonstrating at Linuxfest Northwest this year. I will be presenting on Open Source Telephony and running a demo lab of FreePBX.

I wonder which features I should show off?

New botnet is surfacing

2008-04-07T08:35:00.000-07:00

Gabcast! The VoIP Experience #58 - Zombie warning about Kraken as a new botnet is surfacing

This is a VoIP Experience news alert.

There appears to be a hard to detect botnet on the loose. It is called Kraken.

Google it and educate your users about this.

Aastra and the Power of the Open Telephony Training Seminar

2008-03-27T15:01:00.000-07:00

Gabcast! The VoIP Experience #57 - Aastra and the Power of the Open Telephony Training Seminar

I am interested in working more with XML. During this process I signed up to be a reseller and during the Aastra Webinar I noticed and spoke with several OTTS Grads. It was really odd seeing all these names I recognized popping up in the user list of the Webinar.

It was kind of like old home week... One of the comments was that so many OTTS attendees showing up in the Webinar demonstrated the effect Mark Carson of Aastra had on us.

So, in my opinion, the value of OTTS and the participation of vendors has value beyond the seminar.

Open Telephone Training Seminar - After Hours

2008-03-23T10:35:00.000-07:00

During the recent Open Telephony Training Seminar in wonderful Charleston, SC the attendees, vendors, presenters and organizers all seem to stay together after the "official" class time was over. In fact, since the event was at an Embassy Suites many of us had breakfast, lunch and dinner together.

The host of the event, none other than Ward Mundy of Nerdvittles and PBX in a Flash fame helped us secure some of the finest dining experiences to be had.

Again, since it was an Embassy Suites we all generally attended the "Managers Reception" and gathered in small groups to talk more about Telephony. I felt a bit sorry for Philippe as he tended to get surrounded in a hurry.

I found it amusing how many bystanders ( other hotel guests ) kept asking who we were. At one point a group of ladies informed me that a real gentleman had just "held court" and they had the most fascinating discussion. I don't know what they talked about. What I do know is they had just sat down with the CEO of Sangoma.

After the free drinks each night we set out as a group and descended on some very fine restaurants. The first night we went to the Mellow Mushroom for dinner. The pizza was really good and the atmosphere was fairly psychedelic.

One the second night we made our way to the Boathouse on East Bay. When we set out by foot the large group broke into smaller groups of 10 or 12. Our group was a bit navigation challenged. It is pretty bad when you have to break out a map to go a few blocks :)

Charleston does indeed live up to the challenge of southern hospitality. It was demonstrated by the cheerfulness of the wait staff at the Boathouse as 40 or so of us would walk in. Sure, they knew we were going to show up at a particular time... But still.

In general it can be said that I am a pick eater. I like a really wide variety of foods... but it needs to prepared well. The Boathouse had some of the best prepared food I have had in a long time. Right on par with one of my favorite restaurants.

The next night is was off to 39 Rue de Jean which is on upper King Street... The 39 Rue de Jean is an Ameri-French style place that seemed really popular. It was here that I made good on a small contest I held for attendees and paid for the winners dinner. But, someone picked up my dinner... Crazy. We were a sharing bunch that night as several people bought appetizers for the whole table. I was barely hungry by the time dinner arrived. On the right you can see Philippe and Jared (Digium) engaged in a conversation that sure seemed to have Ward's interest. Overall, it was a pretty nice place. I'll have to give the Boathouse a leg up on food...

On the last night went to Sticky Fingers which is an excellent BBQ joint. We had to wait outside for a little while to get in and that didn't escape the wait staff... they brought us a big plate of hot wings and napkins. The wings combined with the good laugh we were all having made that the best wait-outside-stint I've ever had. Naturally, the food was really good. As a bonus they have some very funny slogans on their souvenir t-shirts and I brought one home for my son.

On one night the organizers did have a private reception which was nice. The drinks were some kind of strong... When the barkeep added the Coke to my Rum it didn't change color... Needless to say I only needed one.

If I missed anything please be sure to comment. One of the interesting things about this event is that it was ran open source style... like the FreePBX project with an emphasis on community.

My hat is off to the FreePBX/Schmoozecom team for extending that community from online to live and in person.

FtOCC vs. OTTS

2008-03-22T15:32:00.000-07:00

Gabcast! The VoIP Experience #56 - FtOCC vs. OTTS

In an interesting twist of fate a fellow enthusiast named Tom Conklin attended the same two inaugural trainings. One was the Fonality trixbox Open Communication Certification and the other was the Open Telephony Training Seminar from FreePBX and Schmoozecom.

This show is Tom and I comparing FtOCC with OTTS.

How did the Open Telephony Training Seminar turn out.

2008-03-08T17:54:00.001-08:00

Gabcast! The VoIP Experience #52 - How did the Open Telephony Training Seminar turn out.

This is an interview with Philippe Lindheimer after the Open Telephony Training Seminar. We discuss how it went and what lessons were learning and are going to be applied in the next OTTS. The next training event is in May 2008.

Can't we all just get along....

2008-02-29T09:01:00.000-08:00

The answer is of course we can. At the Open Telephony Training Seminar we find that many of the vendors attending a getting along just fine. Representatives from Digium(Jared Smith), Sangoma(David Mandelsten) and Rhino Equipment(James Finstrom) are seen here being very chummy indeed. They were not the only vendors supporting the show. i9technologies, Aastra, Snom, and other have shown incredible support for the seminar and we think this is a great sign of things to come.

Three Reviews on the Way

2008-02-18T09:34:00.000-08:00

As I am getting ready for the Open Telephony Training seminar ( which consists of getting house chores done ) I also am preparing three reviews:

1) The Intertex Surfinbird IX67 -Router, Firewall, SIP Proxy and Wireless AP device that was sent over from ABPtech.

2) The 2N VoiceBlue Enterprise GSM to Voip Gateway.

3) The Aretta "PBX in a Flash" hosted solution.

Somewhat of a tech overload as I also plan to blog the Seminar as it going to be ground breaking. So, we will see how quickly I can finish my evaluation. At this time I have the VoiceBlue attached to an Asterisk 1.4.18 system and have inbound calls working. It is an interesting device to say the least. I am going to try a VMWare PBX in a Flash on a MacBook Pro and using the ethernet port connect a Polycom Phone and the VoiceBlue Gateway to the Polycom and see if I can have a stand alone PBX with no Internet or PTSN. Just GSM.

Zimbra's, Yahoo's and Microsoft's... Oh my.

2008-02-03T09:08:00.000-08:00

About a six months into the VoIP passion bug, I was looking for an enterprise level messaging for a large school district which has a 80/20 Mac/PC mix. So, I put the question out to the world... well, the ubergeek world anyway: Mac Calendaring Solutions ?

A fairly good discussion was aroused and out of it I chose Zimbra to investigate and ultimately deploy a few months later. I am still administering this deployment of Zimbra 5.0.1 and of'course still the one who "chose."

What is VoIP about Zimbra? There is an ( as far as I can tell ) unfinished Zimlets[read plug-in] connector for Asterisk. It seems development stopped. Sad and not really the story here.

Why should I be concerned about choosing Zimbra?

Well, because I got some funny looks when Yahoo bought Zimbra. I have no choice but to reassure that Yahoo is not Microsoft and our little Exchange replacement "Zimbra Collaboration Suite" is safe.....

The other shoe dropped and it sounds like I wasn't the only one to hear it hit the floor. Reprinted without permission is the email from Satish Dharmaraj, Zimbra Co-Founder:

Dear Robert,

I am writing to communicate and discuss the unsolicited proposal that Y! got from MSFT. This is a proposal that our board is considering and this may or may not be completed. This is a very early unsolicited preliminary proposal and even if the board did decide to go in that direction there is a long road ahead.

At Zimbra, nothing has changed. We will remain utterly and thoroughly committed to making our customers successful. We will continue to innovate at a great pace. We will continue to disrupt the market with products that are years ahead of our competition. We will continue to embrace open standards. That's our DNA. And we remain committed to doing that.

Many businesses, partners, Higher-Eds and ISPs have committed to Zimbra by becoming paying customers. And there is an even larger set that is part of our open source community. We just made an irrevocable contribution of Zimbra 5.0 to the open source. No one can take that away. We will remain true and faithful to our contractual obligations and will do everything possible to make sure that our biggest assets - you - remain confident that we are here to stay and execute. Thankfully, I can confidently say that the Zimbra movement is bigger than any one company.

We are going to raise the bar exceptionally high next week with the announcement of Zimbra 5.0 and the release of Zimbra Desktop which consolidates many different mailboxes into one user interface. It's the first client-only play by Zimbra where we allow you to use our client to talk to gmail, yahoo! mail or any IMAP/POP source. We are pretty excited about this because it will be ages ahead of its other desktop companions while also being open source.

Thank you so much for allowing us to serve you... we are not planning to stop serving you. I will pro-actively communicate as and when I hear something official and tangible.

Satish Dharmaraj
Zimbra Co-Founder

Satish, all I can say is I hope you had an out in your deal with Y! because I doubt MSFT is going to want to keep an Exchange Killer alive.

For me the great irony of all this that in a cost saving move this school year I deployed "Google Apps for Education" for HS students use. I partially justified this move by suggesting that it would give the Instructional Technology department familiarization with Google Apps in case things go south with the current deployment.

So, not wanting to deploy Exchange had us deploy Zimbra, which was bought by Yahoo and if Microsoft buys Yahoo... is that going to drive us to Google?

I guess I need to prepare for more odd looks when I return to work Monday...

Intermission. Space Age VoIP

2008-02-01T17:09:00.000-08:00

Gabcast! The VoIP Experience #50 - Intermission. Space age VoIP via OTTS

Thats right... Space Age VoIP for attendees of the Open Telephony Training Seminar who will walk away with phone technology that will allow for call initiation and in call voice commands. DUNDi also gets a bit easy to use via FreePBX. This was recorded with a WIP330 as a test.

Some links of note:

Get Official FreePBX Training

PBX in a Flash Forum

Asterisk® Commercial Viability

Philippe Lindheimer Interviews part two-FreePBX 2.4

2008-01-21T13:33:00.001-08:00

Gabcast! The VoIP Experience #49 - Philippe Lindheimer Interviews part two-FreePBX 2.4

In this segment, Philippe Lindheimer-Lead Developer on the FreePBX dev team talks about the FreePBX 2.4 Beta.

Philippe Lindheimer Interviews part one-OTTS

2008-01-21T13:31:00.000-08:00

Gabcast! The VoIP Experience #48 - Interview with Philippe Lindheimer

This is the 1st of a 3 part interview with Philippe Lindheimer the FreePBX Project Lead, In this section we talk about the recently announced FreePBX Training in February 08.
http://www.freepbx.org

Possible FreePBX course offering

2007-12-07T17:02:00.001-08:00

Gabcast! The VoIP Experience #45 - Possible FreePBX course offering

It looks like the FreePBX team is brewing up some training. I would really like to see them offer a FreePBX centric course.

From the post this will be a distro agnostic conference and would be a chance to spend some time with the developers.

This is especially important as the FreePBX 2.4 beta is about to kick off. My guess is that the course will be timed to coincide with 2.4 being in production. Should be an excellent opportunity to get an "under the hood" view of the changes and enhancements.

Lets help make this happen. So, with that in mind please show your support and/or interest at the following link.

FreePBX Course

2007-06-23T18:25:00.000-07:00

Gabcast! The VoIP Experience #37 - FreePBX, JoIP, How I am doing this and Sophos.

This is a short broadcast about the above topics. Really just miscellaneous ramblings on the weekend.

Secure Content Solutions

FreePBX and JoIP

FreePBX, Zaptel 1.2.18, Softphones, Gaboogie, Google Apps

2007-06-10T07:48:00.000-07:00

Gabcast! The VoIP Experience #36 - FreePBX, Zaptel 1.2.18, Softphones, Gaboogie, Google Apps

FreePBX, Zaptel 1.2.18, Softphones, Gaboogie, Google Apps are discussed.

FreePBX® 2.2.2 Maintenance Release

2007-06-08T18:58:00.001-07:00

Gabcast! The VoIP Experience #35 - FreePBX® 2.2.2 Maintenance Release

A Maintenance Release for FreePBX today.

2007-06-03T16:44:00.000-07:00

Gabcast! The VoIP Experience #34 - FreePBX Tutorials, the FCC and selling OSS systems

There are some new FreePBX tutorials to talk about along with a recent FCC bit that is going to include VoIP and makeing an OSS Sale.

FreePBX

FCC's 225 and 25 5

A push to start using Asterisk 1.4

2007-05-31T20:27:00.000-07:00

Gabcast! The VoIP Experience #33 - A push to start using Asterisk 1.4

Out of a developer conference comes the news that 1.2 will be security release only on 1st of August 2007.

Asterisk 1.4 and FreePBX

From SineApps

Updated - Tom Keating, Nortel and Fonality OH NO!

2007-05-24T19:53:00.001-07:00

Gabcast! The VoIP Experience #26 - Updated - Tom Keating, Nortel and Fonality OH NO!

Fonality Re releases a press release...

Tom Keating, Nortel and Fonality OH NO!

2007-05-22T19:48:00.000-07:00

Gabcast! The VoIP Experience #25 - Tom Keating, Nortel and Fonality OH NO!

This is a short narrative about Tom's blog:
"Nortel Strong Arms Open Source Vendor"

See the Post.

2007-05-21T14:48:00.001-07:00

Gabcast! The VoIP Experience #22 - Vyatta and Digium® Inc., team up

http://www.vyatta.com/about/pressreleases.php?id=37

Vyatta and Digium® Inc., team to provide? I hope it is an easier integration for Firewall/Routing and VPN for Voip.

Last Gabcast

2007-05-17T18:36:00.000-07:00

Gabcast! The VoIP Experience #21 - A Little Catch Up

I thought I'd better catch up a bit. FtOCC, trixbox 2.2, AsteriskNow, Asterisk 1.4, trixbox appliance and FreePBX are the topics.

This is a start

2007-05-17T18:31:00.000-07:00

I am going to move my blog from wordpress to here as a trial...lets see how it goes.

Some links from my latest Gabcasts:

FreePBX
trixbox

Asterisk