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AstriCon Asterisk Conference Soon

September 18, 2009 10:01 AM | 1 Comment
astricon.jpgAfter ITEXPO's resounding success in Los Angeles (over 6,000 attendees), we can definitively say VoIP hasn't been as badly affected as other industry sectors within the U.S. economy. In just about 3 weeks, we can confirm this is true with the big AstriCon event held at the Renaissance Glendale Hotel & Spa in Glendale, Arizona. TMC's Internet Telephony Magazine is a media sponsor for the event. Companies participating include Aastra, Adhersion, Digium, PIKA, Polycom, Sangoma, Xorcom, and more

This marks AstriCon's sixth year as the official conference for Asterisk, the world's leading open source PBX. According to the event organizers, "AstriCon's mission is to expand awareness and knowledge of Asterisk over the course of a three-day conference and exhibition. AstriCon includes a wealth of information for every Asterisk user, whether you are getting started or have already discovered the power of Asterisk."

I for one am a little sad I won't be going. Too much stuff to do back at the home office. TMC is growing like gangbusters and we are very close to moving into a state-of-the-art facility. Now imagine you are in charge of moving TMC's entire data center to this new facility with minimal downtime. It's enough to keep any CTO awake at night. It's not happening for a couple of months, but this will require some massive planning by me and my team.

Skype for Asterisk Launches

September 1, 2009 11:10 AM | 1 Comment
skype-for-asterisk.pngAt TMC's ITEXPO, Digium and Skype announced the official launch of Skype for Asterisk, which was launched as a closed beta back in September 2008. Well, now anyone can now download Skype for Asterisk and make & receive low-cost calls leveraging Skype.

According to Digium, "Now businesses can take advantage of Skype's low-cost calling to landlines and mobile phones and free calling to more than 400 million registered Skype users around the world. Skype for Asterisk allows businesses to access the world's largest community of people communicating over the Internet, natively encrypts all voice calls and lets companies manage their Skype user accounts via Skype's Web-based Business Control Panel. Businesses already using an Asterisk-based phone system can add Skype as another complementary form of communications by downloading Skype for Asterisk, without additional costly hardware. Skype users can benefit from the advanced call features of Asterisk, including call transfer, interactive voice response, automated call distribution, flexible call-routing and many more."

"Digium has been using Skype for Asterisk for the past few months while the product has been in development," said Danny Windham, CEO of Digium. "We created Skype accounts such as Digium Sales and Digium Support--a convention I suspect many companies will quickly adopt. Now, our customers all over the world can call us for free using Skype and our Asterisk PBX processes the inbound call just like it would a normal call. This is going to save Digium and our customers a lot of money."

DATUS Corporation is a Digium Select Partner with nearly four decades of experience designing and implementing communications networks in Germany. The company has nearly completed an Asterisk installation at 2,100 sites for LVM Versicherungen, a major insurance firm, and also works with Digium to design features for Asterisk that are of particular interest to European businesses. "Adding Skype for Asterisk to the DATUS indali OBX, our IP-PBX, will offer our customers inexpensive and secure international calling that, for instance, could be used for toll free customer services," said Jonny Kueppers, vice president of sales and marketing at DATUS. "We believe that the price and cost savings will be welcome with today's budgets."

"The combination of Skype and Asterisk gives those companies that have relied on Skype the advanced call management capabilities of Asterisk, while Asterisk users get free calling to more than 400 million registered Skype users and low Skype rates when calling landlines and mobiles," said Stefan Öberg, vice president and general manager of Skype for Business. "We believe the product will bring together two of the largest groups of users that value flexibility and cost savings in their PBX systems."

Foehn Ltd, Digium's U.K. Solutions Partner, has been designing and implementing Asterisk-based solutions for more than five years. The company's technical director, James Passingham, commented: "With Skype for Asterisk, we can offer our clients even more freedom in business communications. The ability to unlock the lower call costs of Skype provides a huge savings opportunity, especially for those with offices and customers around the globe."

Skype for Asterisk Features
Skype for Asterisk, which is compatible with the free and open source Asterisk versions 1.4, 1.6 and AsteriskNOW™, as well as the commercially licensed Asterisk Business Edition™, is unique in the market today. It is the only solution that integrates directly with Skype, enables multiple concurrent Skype calls from a single Skype account, and supports both G.711 and G.729a calling.
  • Make Skype-to-Skype calls.
  • Receive calls with online numbers (SkypeIn).
  • Make world-wide PSTN calls to landline and mobile phones (SkypeOut).
  • Make and receive multiple concurrent Skype calls from the same Skype account.
  • DTMF support for incoming and outgoing calls.
  • Read Skype profile fields from incoming calls.
  • Set and retrieve online status.
  • Set privacy settings.
  • Handle incoming Skype calls using Asterisk applications such as voicemail, ACD, MeetMe conferencing, etc.
  • Simultaneous access from both Asterisk and the Skype desktop client.
  • Trunk calls between Asterisk servers over Skype.
  • Supports G.711 and G.729 (included) codecs.

Asterisk Training Courses at ITEXPO

August 17, 2009 10:22 AM | 0 Comments
itexpo09.gifCan you believe ITEXPO is just two weeks away? It's also almost September. Where did the Summer go?

ITEXPO, the #1 VoIP conference in the U.S., has several educational tracks you might be interested in checking out. Of particular interest to me are the two separate Asterisk and the Switchvox training courses. As Asterisk's popularity continues to grow, so does its development and complexity. Last year's Asterisk isn't the same as this year's, so it's never too late for a refresher or to learn about the newest features.

I've only seen demos of Switchvox and haven't actually put Switchvox through the full test-drive ringer, so I might want to check out the Switchvox training course just to see what's new and what's different from regular Asterisk. Also can't hurt to learn how to use and manage it since I'd like to review it at some point.

You can check out and register for one or both courses here.
Some interesting news from D2 Technologies about them showcasing their mCUE™ converged communications client with embedded VoIP for Android at OESF Japan. I should point out that D2's mCUE mobile convergence software solution combines a communications user interface with the company's vPort MP VoIP software platform and is targeted towards OEMs and service providers to help deliver integrated Fixed Mobile Convergence (FMC) and Unified Communications (UC) functionality.

google-android.jpgAt OESF they will demo how mCUE can enable VoIP, video chat and other IP communications capabilities in stationary Android-based embedded equipment and consumer electronics devices. With mCUE, these devices can offer premium multi-service unified communications capabilities and deliver simultaneous interoperability with any communication service provider, Instant Messaging (IM) community or social networking platform. All popular communication modes are converged to a single communications user interface (UI), including circuit switched voice (PSTN or cellular), VoIP, Instant Messaging (IM), SMS and video chat.

Full release after the jump...
elektrobit-mid-reference-design.jpg
Smartphones, netbooks, smartbooks, and Mobile Internet Devices (MIDs) while very similar feature-wise, each has their own distinct advantages. Smartphones such as the iPhone have been widely successful, while the MID market has been a bit slow to take off. Elektrobit Corporation (EB), based in Oulu, Finland aims to change that with their new MID reference design that combines the "pocketability" of smartphones with the power of PCs/netbooks since it can run desktop Linux applications. Picture an iPhone that can actually run full version Linux applications such as Firefox, Opera, OpenOffice, Thunderbird, SSH client, and more. That's exactly what you'll get with Elektrobit's (EB) new MID reference design. EB's sleek, media-centric MID reference device takes the power of the PC and makes it pocket-able so you don't have to compromise on mobile capabilities.

I spoke with EB's Vesa Kiviranta Vice President, Mobile Internet Device Solutions, Wireless Solutions BU about their new reference design. Vesa explained that it's based on Intel's next generation Moores­town platform. The reference design includes touch-screen support with multi-touch (cool!) support. It uses the latest 3D and high resolution capacitive sensing touch screen (3.97") powered by EB Touch & Feel technology. Because the screen measures nearly 4 inches (3.97"), it fits into the MID category, while smartphones have screens smaller than 3.9 inches and netbooks have screens larger than 5 inches.The reference design relies on a Linux-based OS with EB's MID UI & Application framework based on QT. It also features EB Navigation Suite with integrated GPS.

It supports two cameras (front & back of phone) so not only can you snap photos, it can also easily support videoconferencing/videochat. I asked EB about support for Skype video chat  and they told me it will indeed be supported in their MID design. In fact, they tested it using the Linux Skype application in their labs. As far as I know, this marks the first time a pocketable mobile phone can perform Skype videoconferencing! Cool stuff! I know many iPhone fans were very disappointed the new iPhone 3GS didn't add a front-facing camera (myself included), which would allow for videochat capabilities. Of course, the current Skype for iPhone client doesn't support video, but my sources tell me it's in the works. It will be very hard to have a videoconference if you have to turn the iPhone around so the camera is facing you, but not the iPhone screen. Basically, they'll be able to see you, but you won't be able to see them - unless you spin the iPhone back around.
Your IP-PBX is one of the most critical pieces of corporate infrastructure. It cannot afford any downtime, which is why the fives 9's (99.999%) of reliability was coined. While Asterisk is a pretty stable open source IP-PBX platform, it it still in its infancy, so it hasn't had the same time that the old 'Big Iron PBXs' have had to reach five 9s of reliability. Then again, many traditional PBX manufacturers have abandoned 100% proprietary hardware and use many of the same standard off the shelf components that are in Asterisk, including motherboards, memory, processors, etc. So the old wives tale that big iron PBXs are more reliable than PC-based PBXs no longer applies.

trixbox-logo.jpg Still, Asterisk and all of its derivatives (trixbox CE/Pro, PBX in a Flash, etc.) have a cult following (of which I'm a member) -- and like any cult, we like to do crazy things, like tweak Asterisk or trixbox in the middle of the work day to see if some newfangled text-to-speech feature will work. Well, with so much tweaking by some Asterisk cultists, something is bound to go wrong, usually at the end of the work day on a Friday when you're driving home, forcing a return to the office or waiting to you get home and SSH into Asterisk to restart the service.

So how do we ensure a more reliable Asterisk platform using an automated tool? Surely there must be a way of monitoring the Asterisk service and if it crashes, automatically restart it, right? Ever second is precious when you're trying to achieve 5 9s of reliability, which equates to 5 minutes, 15 seconds or less of downtime in a year. Or if you want to get really crazy, shoot for 6 nines of reliability (99.9999%) which is 31.536s of downtime per year!

monit-logo.jpg Well, before we continue, you must remember that Asterisk runs on Linux and there are many great monitoring tools for Linux. In fact, for the blog web server you're reading this article on, I'm running a free monitoring tool aptly called monit, which you can get here.  This tool is so easy to use, it should be in any Linux admin's arsenal. I use it to monitor various parameters of the blog server, and if certain conditions are met, it automatically restarts the apache web service.

It got me thinking, "Why not use monit to monitor Asterisk?" Well, here's how to do it!

1) Install monit.
2) Simple way: Run 'yum install monit' or run 'apt-get install monit' Go to Step
3) Compile/Harder way: Go here: http://mmonit.com/monit/download/ and download the .tar file, currently called monit-5.0.tar.gz
4) Untar monit
# tar -zxvf monit-5.0.tar.gz
# cd monit-5.0
Configure and compile monit:
# ./configure
# make
5) Install monit
# make install
6) Copy monit configuration file to /etc/ folder
# cp monit.conf /etc/monit.conf (older versions used monitrc filename)
7) Edit monit.conf & put in your monitoring rules (see examples below)
8) Add monit service to the startup. Red Hat command follows:
# chkconfig --add monit
# chkconfig --level 2345 monit on
# {confirm the run levels}
# chkconfig --list|grep monit

It is super easy it to setup the mail server for notifications and to configure monitoring of processes, files, loads (CPU, memory), and ports. And of course, using monit you can monitor Asterisk, trixbox CE or Pro, PBX in a Flash, and other IP-PBXs that run on Linux.

Here's a snippet from two monit.conf configuration files (one the blog server, the other Asterisk):
###############################################################################
##
## Start monit in background (run as daemon) and check the services at 2-minute
## intervals.
#
set daemon  120 # can set lower if want downtime <2min
set mailserver mail.tmcnet.com     # primary mailserver
## You can set the alert recipients here, which will receive the alert for
## each service. The event alerts may be restricted using the list.
#
  set alert blogalerts@tmcnet.com          # receive all alerts
  set alert anotheremailhere@somewhere.com
  check system blog.tmcnet.com
    if loadavg (1min) > 4 then alert
    if loadavg (5min) > 2 then alert
    if memory usage > 75% then alert
    if cpu usage (user) > 70% then alert
    if cpu usage (system) > 30% then alert
    if cpu usage (wait) > 20% then alert
  check process apache with pidfile /var/run/httpd.pid
    start program = "/etc/init.d/httpd start"
    stop program  = "/etc/init.d/httpd stop"
    if cpu > 60% for 2 cycles then alert
    if cpu > 80% for 25 cycles then restart
    if totalmem > 1300.0 MB for 5 cycles then restart
    if children > 250 then restart
    if loadavg(5min) greater than 10 for 8 cycles then stop
    if failed host blog.tmcnet.com port 80 protocol http
       and request "/monit/doc/next.php"
       then restart
    if failed port 443 type tcpssl protocol http
       with timeout 15 seconds
       then restart
    if 3 restarts within 5 cycles then timeout
    depends on apache_bin
    group server

# Asterisk Monitoring rule
set daemon 30 # Check every 30s
set logfile syslog facility log_daemon
set alert asteriskalerts@yourdomain.com
check process asterisk with pidfile /var/run/asterisk/asterisk.pid
group asterisk
start program = "/etc/init.d/asterisk start"
stop program = "/etc/init.d/asterisk stop"
# Check uptime via Asterisk Manager Interface (AMI) port 5038
if failed host 127.0.0.1 port 5038 then restart
if 5 restarts within 5 cycles then timeout

#Check Veritas BackupExec Agent
check host blog.domain.com with address 192.0.0.6
start program = "/etc/init.d/VRTSralus.init start"
#stop program = "/etc/init.d/VRTSralus.init stop"
if failed port 10000 with timeout 35 seconds then restart
Further, you can even test the SIP protocol, which uses port 5060. The SIP test is similar to other protocol tests that monit supports, however, it allows extra optional parameters.

IF FAILED [host] [port] [type] PROTOCOL sip [AND] [TARGET valid@uri] [AND] [MAXFORWARD n] THEN action [ELSE IF SUCCEEDED [[<X>] <Y> CYCLES] THEN action]

TARGET : you may specify an alternative recipient for the message, by adding a valid sip uri after this keyword.

MAXFORWARD : Limit the number of proxies or gateways that can forward the request to the next server. It's value is an integer in the range 0-255, set by default to 70. If max-forward = 0, the next server may respond 200 OK (test succeeded) or send a 483 Too Many Hops (test failed)

SIP examples:
  check host openser_all with address 127.0.0.1
   if failed port 5060 type udp protocol sip
      with target "localhost:5060" and maxforward 6
   then alert
 
  check host sip.broadvoice.com with address sip.broadvoice.com
   if failed port 5060 type tcp protocol SIP
      and target 1234@sip.broadvoice.com maxforward 10
   then alert

Now that you know how to automatically monitor Asterisk, trixbox, PBX in a Flash, etc. those five nines (6?) of reliability are just around the corner. As the PBX administrator / telecom manager, you will be worshiped by your sales team star-trek-who-mourns-for-adonais.jpg and boss for keeping the phone system up all the time. They will think you an Asterisk God, who will be adored and who shall command great respect and admiration. And none shall mourn for any Asterisk outages.

AsteriskNOW 1.5.0 Released

April 1, 2009 8:45 PM | 0 Comments
asterisknow-logo.jpgAsteriskNOW 1.5.0, which launched as a beta in October 2008, is now available for download at http://www.asterisknow.org/downloads. Of course, existing AsteriskNOW users can simply run "yum update" to update to the latest release. I love 'yum' for Linux systems - it's like Windows Update on steroids, but without the Internet Explorer GUI requirement.

According to AsteriskNOW, here are the notable changes since beta2:
* Updated several packages to latest versions (Asterisk, DAHDI, etc)
* Fixed more permissions issues between Asterisk and httpd/FreePBX.
* Updated to CentOS 5.3 (http://lists.centos.org/pipermail/centos-announce/2009-April/015711.html)
Some big news from Digium. Rich Tehrani met with them yesterday to get the inside scoop. Rich takes copious notes on his iPhone, which he sent off to me to try and write up this news. Alas, I've been pretty busy myself, but I wanted to share Rich's notes below, since there are some good "nuggets" in there.

For instance, from Rich's notes I see that Switchvox 4.0 is on the verge of shipping. But the really big news is that Asterisk has announced the general availability of technical support subscriptions for open source Asterisk. Before if you wanted support from Digium, you had to purchase Asterisk Business Edition. Well, no longer. Now, all of you Asterisk fans out there that try Asterisk tim-toolman-taylor-asterisk.jpg(and think you know what the heck you're doing) but get stuck, can now contact Digium and get some support. No more relying on the Asterisk community to answer you questions. Not that asking the Asterisk community is a bad thing, but if you phone system is down, you can't wait hours for someone to respond to an online posting. This could be a huge revenue-generating opportunity for Digium, which can now monetize the open source version of Asterisk with support subscriptions. I'm surprised they didn't offer it sooner. Maybe they were afraid it would upset channel partners?

Rich's notes:
  • Open Source Subscriptions
  • 2 smb subs
  • And 2 enterprise class
  • Incident based
  • Problem: up to today needed community support or consultant with hourly rate
  • Now annual sub - 3 year 10% discount
  • Can call Digium based
  • Level one - support local hours 12 hours - starting at your 8:00 - 7:00
  • For 5 days a week
  • Buys sub online
  • Available in a month through the channel
  • Get a key, name contact and get details when you call
  • Get incident/case handled
  • Can open via we or phone
  • Find a bug - gets entered in bug tracker
  • Gets handled like any biz edition type of bug
  • Not really SLA like a commercial licensed product
  • Biz edition - now only available as OEM or commercially licensed product
  • They want people to buy the open source - engineering opens up 1.4 and 1.6 - first time Digium provides support for open source asterisk
  • Up till now consultants, etc
  • Open source - people buying business edition for support reasons
  • Now getting open source subs
  • Can now support enterprise class apps
  • In the past - anyone who built a large network - 2 levels of enterprise class support
  • 24x7 - server based
  • Unlimited users
  • Up to 3 names contacts
  • First foray into enterprise from server side
  • Up to 24x7 support
  • Switchvox 4.0 on the verge of shipping
The new Asterisk support services enable companies to leverage the power of open source Asterisk with the confidence that their system will be supported by the very founders of the Asterisk movement. According to the news release, "The support subscriptions provide technical support, hardware replacements and substantial discounts on training programs to enable users to take full advantage of the power of the Asterisk platform."

mark-spencer.jpg "Digium's new subscription services give Asterisk users the best of both worlds--they can download and use Asterisk free of charge, as always, and now they can also call on Digium for technical support when needed," said Spencer. "We think the combo of free and open, with support, is going to appeal to many of our most technical users. The Asterisk community has long been a source of great expertise through online forums, and now we're supplementing that with the ability to call us, 24x7, for access to our Asterisk experts."

danny-windham.jpg Danny Windham, CEO of Digium, said: "As Asterisk gains traction within large businesses, demand for professional support is on the rise. Our deep knowledge of open source Asterisk and total commitment to its development makes us ideally suited to offer these new services. Companies that purchase subscriptions will receive support from the most knowledgeable group of Asterisk experts in the industry. We see this offering as a substantial step forward for Asterisk in the enterprise and a valuable service for companies of all sizes."

Asterisk support subscriptions are bundles of services sold on an annual basis. They include technical and engineering support, consultative services, advance hardware replacement, and discounts on Asterisk training and conference passes. 

Asterisk support subscriptions are available immediately from the Digium webstore at http://store.digium.com and will be available through Digium channel partners in Q2. SMB pricing begins at U.S. $595 per year for support during the subscriber's business hours (8:00 a.m.-5:00 p.m., Monday through Friday); 24x7 support for an SMB begins at U.S. $1,995 per year. Enterprise subscriptions, including 24x7 support, begin at U.S. $3,995 per year. Pricing includes a defined number of servers supported and cases opened per year.

You can read the official news announcement here.

Greatest Linux Command Ever!

March 11, 2009 10:22 AM | 7 Comments
This is the greatest Linux command ever! Definitely my favorite.
find ./ -name \*.html -printf '%CD\t%p\n' | grep "03/10/08" | awk '{print $2}' | xargs -t -i mv {} temp/

linux-penguin-logo.jpgWhat it does is look (find) for files that end in .html uses the printf option to format the 'find' output, then passes it to grep for searching for a certain date, then awk for printing a certain field, and finally xargs for executing a certain command.

Let's break it down...

The printf part within the find command has the format '%CD\t%p\n'.

%Cx = File's last status change time in the format specified by x. x=D. D=date in the format mm/dd/yy
\t = Horizontal tab
%p = file's name
\n = newline

So basically it outputs the file's last status change followed by a horizontal tab, then the filename, and then a new line. But before it outputs it, it sends it to 'grep' which searches the output and only outputs lines with "03/10/09".

Example so far: (minus the awk, xargs and mv commands)
find ./ -name \*.html -printf '%CD\t%p\n' | grep "03/10/09"
Outputs this: (notice the tab to separate the 2 fields)
03/10/09        ./2005/05/index.html
03/10/09        ./2005/03/index.html
03/10/09        ./2005/04/index.html
03/10/09        ./linked-in.html
03/10/09        ./consumer-electronics/samsung-bribery-news.html
03/10/09        ./technology/iptv/index.html

Now send this output into the awk command (awk '{print $2}') which parses it and pulls out the 2nd column/field (hence the tab character), which is the filename, including the path.

Here's the output you now have after adding awk '{print $2}' in:
./2005/05/index.html
./2005/03/index.html
./2005/04/index.html
./linked-in.html
./consumer-electronics/samsung-bribery-news.html
./technology/iptv/index.html

Next, send this output of "exact path + filename" to xargs for execution in the Linux shell.

The "xargs -t -i mv {} temp/" part basically takes the input from the previous commands (files named .html modified on 3/10/09) and moves (mv) them to the temp/ folder.

The xargs command can do anything. So instead of moving the files, I could delete them, run chmod on them, or something else.

It took me awhile to write this command. I've used various methods of finding files on Linux servers over the years, but this one is one of the most powerful.

Definitely a command you should have in your Linux arsenal!

p.s. Here's another tip. If you want to search ALL files (not just .html) then use the following command. Notice the \* and not * for the search. That part got me since I didn't think the * (wildcard) had to be backslashed. Usually when you backslash a character that means you want the 'literal' character specified after the \ (backslash) character. I didn't want filenames with a '*' in it. I wanted the wildcard. That threw me for a minute before I figured it out. Anyway, here's the command:
find ./ -name \* -printf '%CD\t%p\n' | grep "03/10/08" | awk '{print $2}' | xargs -t -i mv {} temp/

VoiceGear SkyBridge at ITEXPO

February 3, 2009 11:10 AM | 2 Comments
voice-gear-skype-gateway.jpg IndustryDynamics' VoiceGear SkyBridge is a Skype-to-PBX product that works with analog and VoIP PBX systems and connects them with the Skype network. I stopped by their booth at ITEXPO to learn more. They told me that SkyBridge supports up to 60 Skype accounts and up to 4 simultaneous Skype channels. They demonstrated the web interface used to provision Skype accounts and other settings. It was pretty straightforward and easy to use. They gave me a CD which is a bootable .iso image that will automatically format your hard drive, install Linux, and a full-functioning version of their software so you can 'try before you buy'. They just limit it to 30s calls.

They mentioned that their Skype gateway can not only interface with IP-PBXs using SIP, but through a partnership with Sangoma Technologies, they can also interface with traditional non-VoIP phone systems. Leveraging Sangoma analog and T1/E1 telephony cards, VoiceGear can integrate with PBXs using FXO-to-FXS connectivity as well as PBXs with T1/E1 interfaces.
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