Category: Asterisk
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Calling all Ruby and Asterisk developers! First Adhearsion Conference in SF Aug 14-15
Continue Reading: Calling all Ruby and Asterisk developers! First Adhearsion Conference in SF Aug 14-15For all of you out there working with the Adhearsion open source telephony framework to easily create communications apps on top of Asterisk using the Ruby programming language…. the first “AdhearsionConf” will be held August 14-15 in San Francisco.Jay Phillips, the creator of Adhearsion, will be in the event and undoubtedly a great amount of hacking on Adhearsion will occur throughout the time.
The exact location and schedule are still being confirmed, but mark the date!
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Adhearsion open source telephony framework has new source code repository
Continue Reading: Adhearsion open source telephony framework has new source code repositoryI’ve long been a fan of the work that Jay Phillips did to create the Adhearsion open source telephony framework and so I was delighted to read today of news of its future. To give some context, Jay first created Adhearsion a number of years back because he was frustrated with how hard it was to create dial plans with the open source Asterisk PBX. So Jay went off and created a framework where a programmer could use the Ruby programming language to very simply create voice applications.Jay went on to team up with Jason Goecke to further develop Adhearsion and then last year Jay and Jason joined Voxeo (my employer) to create Voxeo Labs out in San Francisco. While Jay has since moved on, Jason continues to move Adhearsion forward and announced today that Adhearsion has a new home on Github … and hinted at much greater plans in store. I’m looking forward to seeing what all they may be…
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Tim Panton’s VERY cool demo: Google Wave + Skype + Asterisk + Ibook
Continue Reading: Tim Panton’s VERY cool demo: Google Wave + Skype + Asterisk + IbookOver on Skype Journal, Phil Wolf posted about Tim Panton’s VERY cool demo which he gave at Astricon and then apparently just yesterday at eComm Europe. Tim from phonefromhere.com mashes up Google Wave, Skype, Asterisk (with Skype for Asterisk) and Ibook to make Skype calls from within a Wave, complete with recordings of utterances and, naturally, the ability to have an annotated collaboration session in Wave:Phil quotes Jason Goecke (a colleague of mine at Voxeo) describing how it works:
“it is a Google Wave Gadget with his PhoneFromHere.com IAX2 Java softphone as the client. Then, the IAX2 Java phone connects to Asterisk with Skype for Asterisk installed. Then, there is a server-side element, Ibook, that is breaking apart utterances into individual files. So that as each person speaks, it captures it into its own file. Then, as that happens, a text frame is sent from Asterisk to the softphone with the file details. The gadget then uses some Javascript to embed a link. IAX2 supports text frames.”
Read Phil’s full post for more info and for Phil’s views on what this all means.
VERY cool demo!
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Digium takes on the “fax issue” with Fax For Asterisk…
Continue Reading: Digium takes on the “fax issue” with Fax For Asterisk…I can’t stand fax. I can’t. It’s a technology that I just wish would go away. It kills me that fax is one of the main reasons I didn’t drop my landline in my move. Yet the reality is that fax usage is everywhere… and probably will be for quite some time if for no other reason than the complete and utter simplicity of fax usage. Print out your message, or write your message (you know… that thing we all used to do… take a writing tool (pen, pencil, crayon, charcoal, etc.), grasp it in your hand and make marks on some writing surface…), just stick that message in your fax machine, punch in the number and press Send. It’s hard to get much simpler than that.But the lack of fax has been a barrier to many a premise-based IP-PBX deployment. Everything’s going great… people are looking at all the great things they can do with VoIP and Unified Communications, etc. They are figuring out distributed architectures that are all IP-based. It’s all looking really cool technically and will save money, too. All is going well and then someone asks “What about the fax machines?” And so people wind…
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Voxilla Tutorial – Running Asterisk in a EC2 Cloud
Continue Reading: Voxilla Tutorial – Running Asterisk in a EC2 CloudLong-time readers will know that I have been intrigued for a long time with what we now call “cloud computing” (and have written about it and spoken about it) and also continue to find the world of open source telephony interesting.So naturally when I’m pointed to a step-by-step tutorial about running Asterisk in Amazon’s EC2 cloud, I’m interested. 🙂 It’s a nicely done tutorial and I look forward to seeing what people will do with it. (Unlike Mark Headd, who pointed to the tutorial in a tweet, I won’t be trying it out this weekend, but I will be doing so at some point soon.)
Cool stuff…
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Technorati Tags: amazon, ec2, asterisk, opensource, cloud, cloudcomputing
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Clarifying how Asterisk could possibly be used as a Skype-to-SIP gateway
Continue Reading: Clarifying how Asterisk could possibly be used as a Skype-to-SIP gatewayAfter my post yesterday about “Skype for Asterisk” (and the update post) and the potential it allows for SIP interoperability via Asterisk, I’ve received a few comments that seemed to interpret what I wrote as somehow indicating that the Skype announcement somehow meant that there was new “Skype to SIP” functionality in the “Skype for Asterisk” announcement.
Just to be clear, there isn’t any new “Skype to SIP” functionality in the “Skype for Asterisk” piece announced yesterday by Digium and Skype. None.
It is purely a commercially-licensed software module (which most of us speculate will be a binary software module, i.e. we won’t be able to actually see the code) that provides two-way connectivity from Asterisk to and from the Skype cloud. Skype users can call into an Asterisk system. Users connected to an Asterisk system can call out to Skype users. Users on the Asterisk system can also call to the PSTN (via what was called “SkypeOut”) and receive calls from the PSTN (via what was called “SkypeIn”).
That’s it. That was the announcement yesterday. Period. End-of-story.
However, the point I was making in my post yesterday was this announcement has the potential to turn Asterisk into a two-way…
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More on how “Skype For Asterisk” actually works…
Continue Reading: More on how “Skype For Asterisk” actually works…As per usual, Tom Keating gets us more details on the “Skype For Asterisk” beta program I just wrote about… in his update post, Tom explains how it will work:Well, on an inbound call to your Skype username, both your Skype desktop client rings (if running) and your Asterisk IP phone rings. You can take the call using either your PC’s Skype software or your IP phone. Similarly, if someone calls your SkypeIn number, both will ring. Further, if someone dials your corporate auto-attendant, and then enters an extension number, it will still ring both your Skype client and your regular IP phone.
His post discusses how you can assign Skype names to Asterisk call queues and then includes this intriguing text:
When asked how Skype IP-PBX gateway appliances are affected by this announcement, Stefan Öberg VP & GM Telecom for Skype said, “The appliances that are out there now have built their solutions on standard Linux client. They’ve used the public API on that and basically are running many instances of Skype Linux client. Obviously, that’s not the way the Linux client was meant to be implemented. So those solutions are not scalable or reliable to the extend that…
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Does “Skype for Asterisk” tear down some of Skype’s walls? (and allow SIP-to-Skype?)
Continue Reading: Does “Skype for Asterisk” tear down some of Skype’s walls? (and allow SIP-to-Skype?)Does today’s announcement of a beta version of “Skype for Asterisk” signal a way to tear down some of Skype’s walls? And does this move Skype along toward better SIP interoperability?The announcement happened out at Astricon today and TMC’s Tom Keating had one of the first posts about it – updated with info from TMC reporters who are at Astricon. Both the Digium news release and the Skype blog post highlight these four points that Asterisk users will be able to do:
- Make, receive and transfer Skype calls with multiple Skype names from within Asterisk phone systems, using existing hardware.
- Complement existing Asterisk services with low Skype global rates (as low as 1.7€¢ / 2.1US¢ per minute to more than 35 countries worldwide).
- Save money on inbound calling solutions such as free click-to-call from a website, as well as receive inbound calling from the PSTN throughcreate virtual offices all over world using Skype’s online numbers.
- Manage Skype calls using Asterisk applications such as call routing, conferencing, phone menus and voicemail.
I want to focus on one part of the first bullet. Recall that in my last post about Skype and SIP interoperability I talked about how Skype currently…
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Has Asterisk NOT “crossed the chasm” for developers? (Key links to read for open source)
Continue Reading: Has Asterisk NOT “crossed the chasm” for developers? (Key links to read for open source)Jay Phillips is frustrated. He passionately wants to see open source telephony enjoy success all around the world. Yet right now, when people think “open source telephony”, they almost always think of Asterisk… and Jay sees too many challenges for developers embracing Asterisk. Jay, the creator of the Adhearsion telephony framework for Ruby, has spoken about this at recent conferences and pulled together his thoughts in a lengthy post earlier this week entitled “What We’re Not Admitting about Asterisk“.Jay argues that Asterisk has not crossed the proverbial chasm for developers and outlines some of the issues he sees.
What is perhaps most interesting about Jay’s post is the equally lengthy response by Asterisk creator Mark Spencer. Mark responds to Jay’s various points and in doing so provides some good insight into his views on Asterisk’s connections to developers, APIs, etc., as well as the differences between the markets that Digium, the company, goes after versus the “market” of Asterisk, the raw telephony platform.
Both Jay’s article and Mark’s response are definitely worth reading. I’m friends now with both of them and they both bring immense passion and energy to the world of open source telephony. Ultimately they…