Warheads Stacked in the Kitchen
Jason Goecke has a long history in telephony. His unique perspective is built on extensive business experience around the world as well as technical depth. His passion for disruption manifests itself in his on-going support of open source telephony, with the latest phase being his commitment to the Adhearsion project.
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Behavior Driven Systems Monitoring for Telephony
Using C#/.Net to Invoke the Adhearsion API via REST
Recently an Adhearsion community member, David Lawal, built upon the PHP/REST example I posted back in February. David comes from a C#/.Net background and was beginning to learn Ruby in order to use Adhearsion. Then, in his own words:
"...I needed to leverage 'all the work' done in .Net...As I discovered, the marriage of .Net and Adhearsion was lovely, thanks to the RESTful RPC [API of Adhearsion]"So David created an example and has graciously shared it with the rest of us. The example includes a component in Adhearsion that has this method extending the 'methods_for :rpc':
Adhearsion Website and Sandbox Migration Complete
Click to Call on stage with Sinatra, Rack and Passenger
I have been preparing an Amazon Machine Image (AMI) with Adhearsion, Asterisk and all of the available components installed and pre-configur
ed to be made public soon. For the AMI I decided to get the Sinatra web app included with the Click To Call component up and running as a daemon.
I used a combination of Rack and Passenger, since they make the process so easy and production ready. To this end I have updated the README and Click To Call component I covered in my previous blog post to include the how to and Rack configuration file required.
To get this up and running do the following:
- Install the Apache Webserver
- sudo gem install rack
- sudo gem install passenger (follow instructions here for compiling the Apache module)
Skype Rates and Least Cost Routing
Now that Skype is coming to the enterprise with Skype for Asterisk and Skype for SIP, they will need to enhance the data available for their calling rates. Enabling Least Cost Routing (LCR) is a must for any VoIP provider to the enterprise. LCR allows a phone system to determine, on a call by call basis, which VoIP provider to use based on the best rates associated to the country code or prefix being dialed.
As of now Skype publishes a web page of calling rates based on the country name and the per minute rate including or excluding the tax. A few additional items are needed to make this usable for LCR systems:
- The associated country code for each country (ie - '34' for Spain, '1' for the US, etc)
- More granular prefixes where calling rates may differ (ie - '346' for Spanish mobiles, '336' for Frech mobiles, '1212' for NYC, '1712' for Iowa, etc)
- Billing intervals
- A file download in CSV, or similar format, for import into LCR systems
VXML Style Voice Form for Adhearsion
The Adhearsion developer community continues to create new applications and components.
And even more importantly, sharing them. The latest component addition to the line-up is a VoiceXML style Voice Form plugin for Adhearsion, created by @adzap and posted to GitHub.
You may new create your own compenent, include the Voice Form module and create extensive voice menus to your callers. Here is the classic Adhearison Simon example re-factored into a Voice Form:
[sourcecode language='ruby']
class SimonGame
include VoiceForm
voice_form do
setup do
@number = ''
end
field(:attempt, :attempts => 1) do
prompt :play => :current_number, :bargein => false, :timeout => 2
setup do
@number here. Enjoy and many thanks to @adzap!
Adhearsion Developer Community Writing Apps for America
Skype for SIP == Skype for Asterisk DOA?
Today Skype announced Skype for SIP (SFS). Put simply, enterprise telephone systems may now interconnect with the
Skype network to receive calls from the Skype network and place calls to SkypeOut. All without the need to install any special hardware or software on most modern enterprise phone systems (IP-PBXs to be more specific). Skype's new enterprise targeted connectivity uses SIP, the industry standard for VoIP interconnection. SIP already powers the bulk of Skype's revenue, via SkypeIn/SkypeOut, so this is a logical progression to take advantage of the large scale infrastructure already in place at Skype.
This is a tremendous move by Skype and one I have contended for years was necessary for them to make headway in the enterprise. I applaud this step. There are plenty of great posts out there covering this already, including the one by @danyork on Disruptive Telephony.
What does this mean for Skype for Asterisk (SFA) announced last September? At best the value of SFA has been signficantly reduced by this announcement.
Previously SIP interconnection to the Skype cloud was given to the rarified group of larger players such as Voxeo, Tellme, Genesys and others. SFA was the first time this access was going to be brought to the world of open source telephony developers through Asterisk. This provided an immense opportunity for the Asterisk developer community to create new applications to take advantage of this, which lead me to invest time to participate in the closed beta for SFA still underway.
The SFS announcement this morning has just marginalized SFA to applications that benefit from direct dialing of Skype users from Asterisk and from basic presence updates from the Skype network. Gone are the benefits of providing Skype/SkypeIn inbound calls to the enterprise, SkypeOut trunking, etc. More so, SFA is at a disadvantage since you will have to pay a per channel (simultaneous call) license fee on top of any SkypeIn/SkypeOut costs. Further, I suspect that the number of SFA channels available to a single account will be limited for the same reason that SFS does not do SIP to Skype dialing, so that no one may provide large scale alternatives to SkypeIn.
All of this has really taken the wind out of the SFA sails before it even had a chance to make it to a public beta. Digium must now look to quickly add new features. Such as advanced presence information, instant messaging, the SILK codec and others, if they hope to salvage their own investment in the development of SFA to date. While I understand these things take time, the lethargy of getting the SFA to market does not bode well for rapidly trumping the SFS announcement.
Time will tell.
Three new ways to access the Adhearsion Sandbox
Since the launch of the new Adhearsion website earlier this year we have made a Getting Started Sandbox available to developers. We want everyone to easily give Adhearsion a try without having to setup an entire telephone system to do so. All you need to do is install Adhearsion (and maybe Ruby, depending on which operating system you use) and you may immediately begin using our free Sandbox in the cloud.
Here are the new access methods that we have recently added (more details are available on the Getting Started page here):
Phonefromhere has a Java IAX2 client that is a great way to access the Sandbox from the internet. Tim Panton of Phonefromhere has graciously made his hosted system available to us. Check out their app, as I am sure you will find it useful for your new Adhearsion applications as you roll them out.
We have been participating in the Skype for Asterisk (SFA) closed beta. We have already developed the Skype Utils Adhearsion component that I mentioned in a previous blog post and will be making available as part of the Sandbox soon. In the meantime you may use Skype to access the Sandbox to get a preview of how you may easily develop apps for the Skype world leveraging Adhearsion and Asterisk.
Voxbone has kindly donated a 10 channel inbound phone number for use with the Adhearsion Sandbox. So now you may simply use any old telephone to test your Adhearsion applications on the Sandbox infrastructure.
Enjoy!
