Showing posts with label Adhearsion. Show all posts
Showing posts with label Adhearsion. Show all posts

Friday, June 17, 2011

AdhearsionConf 2.0 Is Coming To San Francisco In October.

AdhearsionConf 2.0 Is Coming To San Francisco In October. http://snapvoip.blogspot.com/
If you are an Adhearsion fan, user or just want to learn about it, this is a golden opportunity for you.
The AdhearsionConf 2.0 will take place at the the offices of NextSpace SF, on the corner of 2nd and Market, in the midst or thick of things in San Francisco. The conference will be held on the weekend of October 8th - 9th.
This is what Mojolingo team (bklang) had to say about the conference;

"The speaking agenda is still being organized, but some topics we know will be covered:
  • Advancements of the Adhearsion framework since last year
  • Adhearsion on the ground and in the cloud (premise vs. cloud)
  • Talks from several people who have interesting Adhearsion installations
  • An exciting announcement that will scale Adhearsion, and Ruby telephony in general, far beyond its current state

We are still looking for speakers, so if you have something you want to share with the community, please fill out our speaker response form.

If you are involved in, or even just curious about Ruby telephony, you need to be at AdhearsionConf. Registration will open soon. Stay tuned!"

Mojolingo

Thursday, October 14, 2010

How To Install And Get The Best Of AGItate.

Installing #AGItate http://snapvoip.blogspot.com/
A few days ago, we wrote about AGItate, "AGItate, Asterisk AGI On Tropo" on taking Asterisk to the cloud. Even though it was not very difficult to understand, I wished for more information on AGItate, Tropo, and Asterisk Interface Gateway (AGI) and my wish was answered today. Actually it was answered a few days ago but only now I found it.

Adhearsion team has posted a screencast series on AGItate;

Get your Asterisk server on the cloud!

Friday, October 01, 2010

AGItate, Asterisk AGI On Tropo

#AGItate, Asterisk AGI On #Tropo http://snapvoip.blogspot.com/
Jason  Goeke infromed us today us about the Tropo AGItate via a Tweet today. I have been reading a bit here and there about it but today it really hit me.
Asterisk AGI, Asterisk Gateway Interface is an interface for adding functionality to Asterisk with many different programming languages. Perl, PHP, C, Pascal, Bourne Shell or what ever you chose.
Tropo make it very simple to VoIP, Messaging or collaboration application using web and communication technologies together with Tropo cloud API with Tropo scripting to take your application to the cloud.
Now that Tropo supports Asterisk AGI, now you can make Asterisk applications that can span the cloud, as far or wide as your heart or the app requires. Just like the cloud PBX, OpenVoice.
From a blog post that Jason made today, key features of AGItate are;
  • Tropo can now speak AGI over TCP to any FastAGI server, including Adhearsion, PHPAGI, Asterisk-Java, etc.
  • A long list of Asterisk commands are supported, and of course all the Tropo ones.
  • While the script is written in Ruby, no Ruby knowledge is necessary to use it. Just point your Tropo application to the Ruby script on Github, upload a configuration file via FTP or WebDAV to your Tropo account, and you are ready.
  • Full support of Tropo Speech-Synthesis (TTS) and Speech-Recognition (ASR), but also for Asterisk sound files. Yes, tt-monkeys works!
  • Fail over to a SIP URI, in case your FastAGI server does not respond. This could be to another Tropo application, another Asterisk box, or anything that supports SIP.
  • You get all of the Tropo channels over AGI, including SMS, Instant Messaging and Twitter, all using the same application.
  • Support for custom SIP Headers in and out of Tropo.
Beyond using AGI to write an entire cloud communications application, Jason tells us that we can also seamlessly integrate Tropo services into existing Asterisk systems using the interface you already know, SIP.
If you want to hear a Tropo AGItate ‘hello world’, give it a ring:
  • Phone/SMS: (408) 641-4410
  • Skype: +990009369991456829
  • SIP: sip:9991456829@sip.tropo.com
  • Jabber/XMPP: tropo_agitate@tropo.im
Tropo AGI

Wednesday, February 25, 2009

Skype For Asterisk Beta (SFA) User Experience.

http://snapvoip.blogspot.com/
Jason Goecke of Adhearsion fame, has been working on Skype For Asterisk Beta (SFA) and I like what he has to say.
"My impression of the implementation so far is good. There is a lot more functionality in the software than I had expected, including support for Skype presence information within Asterisk."
Also the installation and the use of Skype for Asterisk seem to be easy;
"I received the software and was up and running within 10 minutes, as the install process was trivial. I have managed to place calls from a SIP client connected to Asterisk to and from users on Skype. The sound quality is good and interoperability with the rest of the Asterisk features works well."
You can read more at Jasons blog;

Monday, February 02, 2009

Adhearsion v0.8.1 Newer, Better And Shining Like A Ruby!

http://snapvoip.blogspot.com/
I mentioned about Adhearsion, The open-source Ruby framework sometime back. But the Adhearsion and the whole framework behind it, including the code, site and the business has gone very far in that short distance.
Today Jason Goecke was kind enough to remind me of the fact. But when I visited the site, I realized that I need to update my links in the previous article, "The Open Source Ruby Framework Adhearsion" as the Adhersion site has changed a quite a bit.
The team (Yes it is we, not I as Jay Philipps said) is is progressing very nicely and offering a bunch of changes to program, the web site and also adding a sandbox to the Adhersion. So you can stay in SIP world totally ignore Asterisk (hope not) if you want to. As Jay mentioned "The user doesn’t even have to have ever heard the name Asterisk.This is a huge advancement in bringing voice development to the masses and the fact that Adhearsion is able to function this way by just enabling a plugin clearly affirms the potential the new Adhearsion version offers."
I was am very attentive to what Jason is doing to Caller ID with his whitepages plugin.
You can lear a lot more at Adhearsion site and of course from the links above.



Wednesday, December 17, 2008

The Open Source Ruby Framework Adhearsion

http://snapvoip.blogspot.com/
The open-source Ruby framework Adhearsion (What's Not Telling About Asterisk? ) has updated the roadmap and it is very promising. The day that everyone will be able to make their own VoIP application or platform is coming near. For me the configuration file generators with Asterisk has been the high light. But due to time costraints, testing using has been a few jabs rather than actual productive work. The roadmap.
But the roadmap is entising me to pay more attension to the development process and some more added benefits are coming in the form of FreeSWITCH compatibility and YATE Compatibility.
If you are interested, this tutorial might be of help.

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