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Asterisk and Skype™ Together

Source: voip-tech.blogspot.com

Asterisk and Skype™ finally will work together, in the next version 4 of Skype™ will be implemented the code to manage and interface it into the famous and free software Asterisk, software that allow to create a "telephone exchange" for companies with a ridicolous cost.
Will this Skype™ move take the VoIP mastership also in companies, in addition to private users?

For more informations about Asterisk PBX software…
Asterisk official site: http://www.asterisk.org

Published on September 30th, 2008 under , , ,

VoIP Like You Give a Damn

Source: gigaom.com

When I checked out Google’s blog post Tuesday about its Free the Airwaves project, which aims to convince the FCC to approve the use of the white spaces between the spectrum vacated by analog television channels for broadband access, I saw it offered the ability to phone your Congressman. I thought that was kind of cool, so I clicked through to learn more.

I found myself at the master’s thesis of Fred Benenson — a VoIP-based program called Cause Caller that mixes IP telephony and activism. At the site you can enter your telephone number and Cause Caller makes a VoIP call to one of a randomized list of Congressional reps. So far 11 people have made calls on behalf of the Google campaign, which is exactly where things stood on Tuesday when Google provided the link. On the site Benenson said he funds the project himself, so I wondered if an influx of Google calls might bankrupt him, or if Google had volunteered to help offset costs.

Apparently the answer to both is no, and since few calls have been made so far, Benenson may not have to worry. So far Benenson says his most expensive cause has been an effort to impeach President George W. Bush that generated 1,000 calls, but also says he pays less than 3 cents a minute for VoIP and uses Amazon’s EC2 for his servers and Asterisk for the PBX. The EC2 is the most expensive part of the project, which in total has cost him about $500 so far. Benenson has a day job at Creative Commons, so he’s not looking for a revenue model, and says he doesn’t mind footing the bill so far.

“I keep it alive because it’s a fun hobby,” Benenson says. “I basically did the whole site by myself from the design to the VoIP programming, so I kind of took a long hiatus, but now I’m ramping up and starting to blog about it again. The Google notice is like a shot in the arm.”

Cause Caller strikes me as one of the more interesting ways that technology can intersect with politics, with the potential to make a greater impact than emailing petitions and encouraging voter engagement by texting a candidate’s running-mate announcement.

Published on September 5th, 2008 under , , , , , ,

Nokia leaves Asterisk users in the cold

Source: goebel.net

A commentator to my last post "Why Truphone and Gizmo5 applaud that Nokia turns it’s back on mobile VoIP" doubts my argumentation by asking:

I thought Truphone is based on the built-in SIP client? Then it would seem unlikely that Truphone applauds Nokia dropping the mobile VoIP stack from certain models.

My answer is the following:

Yes, Truphone until now works on top of the built-in SIP client. But the Truphone software develops more and more into a standalone application: with the inclusion of SMS, callthrough where no Wifi is available, presence information and so forth. They aren’t afraid of building their own SIP app since it ties the customer even more to them. Therefore Gigaom wrote:

Truphone isnt waiting around for Nokia to do something. A company spokesman told us: From Truphones perspective Nokia has removed the VoIP client from all the N-Series phones for the planned future. We are putting in a replacement client functionality so that existing customers are not orphaned.

Don’t forget that Truphone has a very high pricing for Wifi calls! Their software is convenient to install, but many other VoIP companies are three times cheaper. That’s why they would be very happy to be your only mobile VoIP provider. Vyke already launched their own client, as you can read here, and Gizmo5’s CEO Michael Robertson officially applauded Nokia’s move in a FierceVoIP article.

The only losers are the cellphone users, since these 3rd party apps are much more difficult to use than the native SIP client. Read this insightful comment, posted at Phoneyboy’s blog:

"Im using VOIP on Nokias phone via my own asterisk server. How can Nokia expect me to develop my own Internet telephony application so that I can continue to use it? There are simply thousands of small users out there for whom this is beyond what they could do. This will leave them out in cold.

And further comment. Any third party application will have hard time to match the comfort of integrated symbian UI, where normal and internet calls are integrated together and one push of a button decides which one to make. Just compare this with Fring whose UI is just terrible."

We tinkerers who use Asterisk, Voxalot, Voipstunt, PBXes and Iptel.org are out of the game for the new Nseries devices. I am afraid that the Nokia E71 is the last cool device for a VoIP aficionado like me. Hopefully the Android devices will have more to give. Phoneboy calls us, who use 10 VoIP providers on our Nokia devices, a "minority". Nevertheless he "understands the frustration". Thank you!

But still I think that he is wrong, or maybe just blue-eyed, when he says: "It sounds like the problem is only limited to these two handsets". The problem affects all Symbian Series 60 3rd generation Feature Pack 2 (S60 3.2)! This means: All new handsets from now on are affected. Nokia’s VoIP isn’t revolutionary disruptive anymore, but has changed to a big boys’ only business.

Published on August 31st, 2008 under , , , , , , , , , , , ,

Want To Save $40,000?

Source: asteriskblog.com

trixbox

Who doesn’t? Well, if you really want to save that much, you might want to consider switching over to VOIP for your business. It has long been acknowledged that VOIP services are far more cost efficient than traditional telephony services. This is, in fact, the main reason many individuals and businesses are now opting to make use of VOIP.

For Hail & Cotton, the experience is something to be proud of and share with others. Hail & Cotton is a tobacco distribution company in Springfield, Tennessee. When Tres Ransom, the IT director of the company, was tasked with the overhauling of their telephone system, he was faced with a decision. This was whether to either upgrade what they were currently working with or whether to totally trash the system and go with a new one – that is, VOIP.

As with many IT personnel, VOIP seemed like a very feasible option. Ransom had some concerns, however, as he was not sure of what he was getting into. More so, he was not sure that he could do the work all by himself. Still, he did some preliminary legwork and calculated the costs of each option.

If the company were to simply upgrade their current system, it would have cost them as much $64,000. On the other hand, his researched showed that the company would have to shell out only $25,000 with VOIP. Ransom did his research on Asterisk and Trixbox at this time. As a matter of fact, Ransom did not have to do all the work on his own as he had some outside help from Voice Pulse, a well known VOIP service provider. What happened was that Voice Pulse gave the company a free trial using Voice Pulse’s Connect for Asterisk and Trixbox 2.4 CE. Ransom was quite impressed with the free trial PLUS the fact that the whole package cost almost $40,000 less than his other option.

This is indeed a prime example of what VOIP can do for a business. I think it is important to note the important role of the VOIP service provider as well. If Voice Pulse was not able to provide a clear option for Hail & Cotton and if they were not able to provide that free trial which was what impressed the client, then the deal would not have been possible.

Published on April 25th, 2008 under , , ,

Asterisk Vulnerability Discovered

Source: asteriskblog.com

man hitting computer

Here is something for all Asterisk users out there.  Though we may all be very enthusiastic about Asterisk and the service it provides, we have to be practical and keep our eyes open for vulnerabilities.  Even the people over at Digium do not act like ostriches and keep their head buried in the sand – I guess most other service providers act the same way.  They are always on the look out for weaknesses that other unscrupulous individuals may take advantage of.

Recently, Joel R. Voss aka. Javantea reported a vulnerability in Asterisk systems that may result in denial of service.  Many other sites and blogs have subsequently spread the word about the possible problems that may arise from the vulnerability.  People over at Digium themselves have released an advisory about the issue.  They have also released work arounds that could help solve the issue and avoid potential problems that may arise from it.

Below is the description of the vulnerability as well as other important details that you may need to resolve the issue.  This was taken from Secunia:

Description:
A vulnerability has been reported in Asterisk, which can be exploited by malicious people to cause a DoS (Denial of Service).

The vulnerability is caused due to improper verification of ACK responses during IAX2 handshakes, which can be exploited to spoof an IAX2 handshake and cause a DoS via high bandwidth usage.

The vulnerability is reported in the following versions:
* Asterisk Open Source 1.0.x (all versions)
* Asterisk Open Source 1.2.x (all versions prior to 1.2.28)
* Asterisk Open Source 1.4.x (all versions prior to 1.4.19.1)
* Asterisk Business Edition A.x.x (all versions)
* Asterisk Business Edition B.x.x (all versions prior to B.2.5.2)
* Asterisk Business Edition C.x.x (all versions prior to C.1.8.1)
* AsteriskNOW 1.0.x (all versions prior to 1.0.3)
* Asterisk Appliance Developer Kit 0.x.x (all versions)
* s800i (Asterisk Appliance) 1.0.x (all versions prior to 1.1.0.3)

Solution:
Asterisk Open Source 1.2.x:
Fixed in 1.2.28.

Asterisk Open Source 1.4.x:
Fixed in 1.4.19.1.

Asterisk Business Edition B.x.x:
Fixed in B.2.5.2

Asterisk Business Edition C.x.x:
Fixed in C.1.8.1.

AsteriskNOW:
Fixed in 1.0.3.

s800i (Asterisk Appliance):
Fixed in 1.1.0.3.

Provided and/or discovered by:
Joel R. Voss a.k.a. Javantea

Original Advisory:
Asterisk:
http://downloads.digium.com/pub/security/AST-2008-006.html

AltSci:
https://www.altsci.com/concepts/page.php?s=asteri&p=2

Here’s to hoping that you will be able to take care of the vulnerability before anything adverse happens!

Published on April 23rd, 2008 under , ,

Visual Dialplan for Asterisk On Linux Beta

Source: snapvoip.blogspot.com

Beta release of Visual Dialplan for Asterisk for Linux

Apstel has released their first public beta of Visual Dialplan for Linux. This is in response to tremendous demand for Linux version of popular rapid application development environment for Asterisk dialplan development.

Apstel Dev team is looking for Asterisk users, professionals and enthusiasts who would like to test and provide the feedback on their experience with the Visual Dialplan for Linux. So if you would like to do so, join the program, help to improve Visual Dialplan for Linux.
Because if you do, you might end up having a copy of final product!. Apstel has promised that the five most productive feedbacks will be awarded with free of charge final product license.

Click here to download the Linux version of Visual Dialplan for Asterisk and send us your feedback at betaprogram@apstel.com no later than March 31st, to qualify for the complimentary final product package.

Published on April 8th, 2008 under

VoiceXML Browser For Asterisk Gives Web-enabled Interactive Voice and Video Response (IVVR) applications.

Source: snapvoip.blogspot.com

Digium, the creator of the Number one open source VoIP IP Telephony and IP PBX software Asterisk, has partnered with I6NET to provide an VoiceXML browser as an add on for Asterisk. Asterisk powers Digium’s family of software and hardware appliances including AsteriskNOW™, Asterisk Business Edition™ and Switchvox™.
So what does this VoiceXML browser does to Asterisk? You can talk and interact with asterisk via IVR applications. It will provide the capability to create interactive voice and video applications. I6NET’s add on first complete VoiceXML browser add-on for Asterisk.
VXI* VoiceXML browser for Asterisk gives developers, operators, and service providers the ability to rapidly develop and deploy innovative VoiceXML-controlled voice and video applications in VoIP, PSTN, and 3G-324M networks. VXI* is fully compliant with W3C VoiceXML 2.0 and some 2.1 specifications, and can easily integrate automatic speech recognition (ASR) and text-to-speech (TTS) systems to enable advanced IVVR applications, voice and video communications, and real-time video calling applications. VXI* interpreter is pluggable into standard Digium Asterisk PBX releases including Asterisk Business Edition. The solution provides the important benefits of scalability and a low cost solution profile. Most users of Asterisk can run VoiceXML in the same server.
"Choosing the Asterisk telephony platform is the most important decision we have taken for deploying our VoiceXML browser", said Ivan Sixto, CEO at I6NET. "Asterisk is a strong system on which developer’s ecosystem can easily build advanced multimedia services with VXI*. Running web-enabled Interactive Voice and Video Response (IVVR) applications is now possible on the Asterisk platform."

"Asterisk is designed from the ground up to provide a flexible platform for developing value-added applications," said Jim Webster, director of technology partnerships at Digium. "The work I6NET has done provides a new option with powerful potential for speech solution providers. VXI* is a good example of business-focused innovation from the worldwide Asterisk ecosystem."

The complete press release could be found at Digium Mediacenter.

Published on March 17th, 2008 under ,

10 More Reasons to Love and Like Asterisk.

Source: snapvoip.blogspot.com

I love and like Asterisk for many a reasons and if I am to list those, I will have to pay Google for consuming half the space assigned for Blogger. Linux, which I spend quite a lot of time with to create Grids and clusters. Some of these grids that we create are worlds largest. So Linux gets the first place in my projects/applications/OS list. But right next to it, I have Asterisk, that I also use in various ways.
But Bill Miller (also known as BeeLineBill) from Digium has given 10 thing to know and love about Asterisk. They are similar and different from the reasons that I love and like Asterisk. Sort of like looking through set of funny mirrors. Some aspects are enlarged and some others are amplified.
I think I distorted facts enough and without confusing you further, I will guide you to Bill’s Article. Enjoy a good and informative reading.

Published on March 14th, 2008 under ,

Asterisk 1.4.19-RC2 Is Released By Asterisk.org

Source: snapvoip.blogspot.com

The Asterisk development team released Asterisk 1.4.19-rc2 on 11th of this month. This release, forerunner for Asterisk 1.4.19, which will be made available after a 1.4.19 RC2 goes through testing without finding any major regressions.

The RC2 release contains a number of bug fixes, and one crash regression found during the RC1 testing.

You can download the release as a tarball, as well as from svn.

But most important request is that you download and test this release and report any problems to http://bugs.digium.com/.

Published on March 14th, 2008 under

Free Asterisk Book From O’Reilly, Still Available

Source: snapvoip.blogspot.com

The Free Asterisk Book Asterisk The Future Of Telephony by O’Reilly (released under the Creative Commons license) is still available for download. If you are just starting with Asterisk or want to know about asterisk, there is no better resource.

Published on March 14th, 2008 under ,

Free Calls With SIP Proxies, Asterisk, And A Bit of FreePBX.

Source: snapvoip.blogspot.com

Connect for free, talk forever, pay nothing.
Nerd Vittles (NV) always surprise me one way or another. Today it is SIP Proxies. Last time I looked into Asterisk SIP proxy was when a client needed me to consult on firewalled Asterisk. As the link suggests, I used SIPproxD for that purpose. I think it is still a good solution for firewalling or masquerading an Asterisk Server.
Then there is Dundi, (if you do not know what Dundi is read the Asterisk Documentation or wait till NV writes about it, they promised.
It was easy for me to test the NV demo as were were testing PBX in A Flash both here in USA and a far north corner in Japan. We skipped the kick-ass.net and configured our own DNS servers. Followed the instructions and it worked. (Had to correct my own typos a few times, typing English on a Japanese keyboard is always hard)
But as NV mentioned, what amazed us was the difference in call quality. Now we will have a permanent SIP proxy serving Japan and USA.
In addition, we now have IPKall numbers and we know the phones we use has been selected by NV to be the best IP Phone, the Aastra 57i CT.

Published on March 14th, 2008 under ,

Snap VoIP with Snap At Snapnumbers

Source: snapvoip.blogspot.com

Today I came across snap numbers. It seems that Snap is a dialer and call pop up application for Asterisk. It works by sending the phone number you wish to dial to your Asterisk IP PBX and initiating a call back to your phone, be it VoIP or a PSTN phone. Once your phone rings you pick it up and complete the call. One notable feature is the Microsoft Outlook integration. Outlook users, the call pop-ups are integrated with outlook contacts and the information about the contact will be shown including pictures if they are in the outlook.
In fact it integrates with other applications like Excel, word and powerpoint from Microsoft office. There is a TAPI extension that allows to integrate with application like goldmine, and ACT.
One thing to remember is that this is not a softphone. I would say it is a plugin for your phone, be it a softphone or a regular phone. It extends the capabilities of either phone.
Also the plugin for Firefox and Thunderbird can detect phone numbers on web pages and make them dialable by simply clicking. Something that Google tried with Click-to-call. If they are to populate this concept, they only need a good marketing and PR campaign. Kingmaker at voip watch, comes to my mind.
snapnumber is here. How to configure snap for Asterisk/Trixbox tomorrow.
All in all I am still loyal to my Asteridex fro nerd Vilttles.

Published on March 14th, 2008 under , , ,

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