VOIP Today magazine contributing author Alaa El Fahham spoke today with Mark Spencer about Business , technology and products. It’s a very intersting interview with Digium‘s CTO and creator of Asterisk, the Open Source Telephony Platform. Source: VOIP Today – Exclusive interview for VOIPToday Magazine Mark with Mark Spencer, creator of Asterisk
Around the world, organizations are conscious about IT spending and are continually evaluating ways to slash their operating expenses while keeping up with the latest technological trends. The City of Schoten, a city of 33,000 inhabitants located in the heart of Belgium, recently migrated from 15 disparate telephone systems to an open-source PBX solution to reduce operating and maintenance costs and gain greater functionality to meet their telephony needs. Moving to Digium’s Asterisk, the organization expects to save nearly $50,000* annually while reaping the rewards of a robust IP-based telephony solution. According to our partner Avanzada7, the City of Amsterdam is following Schoten, implementing Asterisk for their telephony too… Here in Spain, we are devilvering many projects for Cities like the City of Avila over Asterisk and VXI* for IVR voice portal services… A Wind of Change is Blowing Through […]
Today a re-worked “asterisk.org” website has been published at www.asterisk.org. We find a new definition about the Asterisk Open Source Software approach, with better design, contents and a new main title : “THE OPEN SOURCE TELEPHONY PROJECT”. We are please to find these new updates; of course, Asterisk was already more than a PBX for its users… This software is the most important Open Computerized Telephony Integration platform, which have a number of very useful input/output telecom channels. I6NET says: Welcome, to this new Asterisk’s community website! Source: www.asterisk.org
Something in changing the Future of Telephony, a Revolution is coming… Asterisk, The Open Source PBX created by Mark Spencer from Digium has enjoyed rapid growth over the past 10 years. As a result, the company has been able to challenge larger, more established competitors to claim more of the market. With Asterisk software, companies can adopt corporate phone systems that are designed to be easier to customize and cost a fraction of traditional proprietary systems. This is only the beginning of the change and it’s first time that Digium is reported by Gartner. All Asterisk’s ecosystem members know that Asterisk is growing fast in the market, like other open source projects (firefox, apache, linux…) have already done. Source: http://mediaproducts.gartner.com/reprints/microsoft/vol6/article15/article15.html
This very insteresting challenge from Digium for next Astricon 2009, seems to have a winner. We are very happy of that great new! The first person to get an Asterisk system moving 10,000 G.711 call legs through a single instance on a single machine will get a first-class steak dinner at Astricon. And a great bottle of wine, if that is your preference. This isn’t an X-prize, but the concept is the same – think of it as an S-prize. ”S” means “Steak”. Or maybe “Salad” if you’re a vegetarian. […] Ten thousand channels sounds like a lot, and it is. But it can be done, and is already done with custom hardware from closed-source vendors. Open Source Asterisk has not been yet tested at anywhere near that high capacity, though attempts have been made in the thousands of channel […]
Thanks to Jim Dalton from TransNexus for these Benchmarks of Asterisk PBX. Most of our customers want to tune their Asterisk Server in order to get the maximum VoIP call performance with VXI* and they need information about the Asterisk kernel behavior during a stress test. We have added our data about VXI* process %CPU utilization. Of course, it’s an average estimation, because VoiceXML applications can execute very different XML dialogs for each call and can require codec translation too; but the VXI* VoiceXML addon for Asterisk use few CPU compare to the PBX for most of voice interactive applications. Diagrams shows % CPU utilization vs Ports (simultaneous calls) – g711 no codec translation TransNexus document describes a benchmark test and performance results for a standard Asterisk PBX 1.4 working as B2BUA (Back-to-Back User Agent). The purpose of this stress […]
We are pleased to announce the open beta of Cloud VXI* VoiceXML for Asterisk is ready to begin and we look forward to you participation. All registred users to this program will receive today the lasted VXI* 4.2 packages to run on al virtual OS ready for Cloud. This software and license keys are only available for users who have previously register to the VXI* Cloud Beta Program to follow any feedback with I6NET’s Support team. Thank you for your continuous support!
All of the more basic DTMF-driven IVR services can be re-created with VoiceXML and the open service architecure that it espouses. The caller isn’t going to notice much of a difference unless you take the migration as an opportunity to rework the user interface. However, moving from a proprietory IVR to a more flexible modern architecure does bring a lot of technical advantages. The advantage that VoiceXML offers is the speed with which changes can be made and new functionality added. You are no longer dependent on your IVR supplier to make any modifications you require and you can re-use existing IT infrastructure and interfaces. These engineering changes have an effect in terms of the product, as any modifications are now easier and quicker to do, resulting in a faster time2market for better customer-oriented services. Actually, by randomising some prompts […]
The dialplan is the routing core of an Asterisk server. Its sole role is to look at what is dialed, and route the call to its destination. This is the core of any telephony system and Asterisk is no different. VXI* VoiceXML browser for Asterisk provide to the PBX a new function called vxml() to extend your favorite telephony system with VoiceXML 2.0+ execution. VXI* provide dynamic XML programming services thru HTTP / HTTPS and using PHP, ASP, JSP, PERL, Python, CGI, C#, etc… to create and run advanced web enabled speech applications on your Asterisk. The dialplan is made up of three elements: extensions, contexts, and priorities. An extension is number or pattern that the dialed number is to be matched against and a context is a collection of extensions (and possibly other included contexts too).