Category Archives: Asterisk

Asterisk

SwitchVox Free Edition

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In any technology solution, the upfront cost, or the initial cost needed even when just want to initially test if the solution will work for your needs is one factor that many business clients consider in their evaluation and purchases. This is true for most solution stack, including PBX/VOIP Solutions.With the advantages and savings that open source solutions such as Asterisk offers, more and more companies feel the need to integrate Internet telephony in their infrastructure. Now, with the latest “offspring” of Digium’s purchase of Switchvox, small businesses can “test the waters” before spending the money on their PBX systems. Thanks to  Switchvox’s Free Edition. It lets potential customers to use the PBX software with zero upfront cost.

Switchvox’s Free Edition is the company’s latest offering, announced a month after it was acquired by Digium, the company behind Asterisk: the open-source Internet telephony platform / PBX on which Switchvox’s product is based. The Free edition joins the other two offerings of Digium / Switchvox: Switchvox SOHO and Switchvox SMB. This feature matrix showcases the differences betweent these three Switchvox offerings.

Switchvox Free Edition is aimed for businesses who want to try and get started with a full-featured phone system that’s easy to administer and use. This can include both existing users of other PBX software and those who haven’t made the jump to IP yet.

With this free edition, 15 virtual, IP or analog phone extensions can be created, and it can handle up to 8 cuncurrent phone calls. This is perfect for small-sized organizations who don’t have a demmand for large extensions.

Switchvox free edition will work with existing hardware, but the Switchvox FAQ suggests that if you want to eventually migrate to the more fully-featured offerings, potential customers can take a look at their Certified Hardware List

Althought Asterisk, one of the core components of Switchvox Free Edition (as well as in other editions) is open source, the whole solution stack is not. Switchvox has closed, proprietary components.

When you have decided to upgrade to SOHO or SMB versions of Switchvox, the company provides a seamless migration path that only requires you to make a payment to the Switchvox store, then you can finish the upgrade using Switchvox’s web console. Switchvox Free Edition is another grate additoin to Asterisk’s growing arsenal of Internet Telephony solutions.

Asterisk Proves Profitable for MyVOIPConsultant

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According to MyVOIPConsultant, an Asterisk Consulting Service, their porfits for the first two quarters of this fiscal year (2007) exceeds their expectations giving them record-breaking profits for the year so far. MyVOIPConsultant co-founder and CEO, Cullen Powell, attributes the success of their business to the growing popularity of Asterisk. According to Powell, “The VoIP industry..has an estimated 10 million VoIP users in 2007 and a number expected to grow to 24 million by the end of 2008, you can get an idea of how well the industry has been treating us. The bottom line is [Asterisk] saves people money, and lots of it. Especially businesses. Everyone is switching to VoIP, and if they’re not they will eventually. You can say goodbye to your conventional phone systems. As for MyVoIPConsultant, we really didn’t expect to see these kinds of numbers and we’re just amazed at how successful the company has been. Watch out for us in 2008, because we’ll be rapidly expanding into new markets.”

The popularity of Asterisk-based VOIP systems comes as no surprise especially since as MyVOIPConsultant website puts it, “99% of businesses with 4 or more land lines who drop their conventional phone service for an asterisk based VoIP system save over 60% on their monthly phone bill”. And that is just on the phone bill alone. Aside from the savings Asterisk, being open source, is very customizable so that it can fit almost any business’ needs.

Although there are no solid reports right now of the profits of other businesses dealing with Asterisk it wouldn’t be surprising to know if they are also having record profits this year. According to Telecompaper, “The global revenue from retail IP telephony services reached USD 6.91 billion in 2006, up from USD 1.83 billion in 2005, according to new research from Point Toptic. Western Europe generated revenue of USD 2.64 billion, followed by the Asia-Pacific with USD 1.75 billion and North America with 2.41 billion revenue. Monthly ARPU was highest in North America at USD 20, followed by Western Europe and SE Asia with USD 15 each and the Asia-Pacific and Latin America with USD 10 per month each.”

Digium’s Partnership Program

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Digium recently launched its new partnership program. The program’s aim is “to form a closer relationship between Digium and companies that have implemented Digium and Asterisk technology in their products.” The partner program is of course beneficial to Digium since it will encourage partner companies, and those that aspire to be partner companies, to develop even better Asterisk-based products. On the other hand, being a Digium partner is also very beneficial especially in terms of product and services promotion.

Digium partner companies fall under five categories: Interoperability Partners, Service Provider Partners, Software Partners, Solutions Partners, and Asterisk Training Providers. Listed below are the description of what each partners has to offer to their clients as Digium describes on their website.

“Interoperability Partners – Interoperability Partners have products that are complementary to and interface with Asterisk. These products interact with Asterisk through a standards-based interface (e.g. SIP) and are certified for interoperability.

Service Provider Partners – Service Provider Partners offer telephony services that can be used with Asterisk. This service may include traditional circuit-switched (PSTN) or packet-switched (VoIP) offerings.

Software Partners - Software Partners offer applications that are used in conjunction with Asterisk to provide enhanced functionality (e.g. a billing platform). Partner software products include those that interface with Asterisk using an Asterisk-specific interface.
Solutions

Solutions Partners – Solutions Partners offer a “whole product” solution based on Asterisk and incorporating Digium hardware and/or software. These offerings may be targeted at specific vertical industries or other niche markets.

Training Provider Partners – Asterisk Training Providers offer high-quality Asterisk training around the world, providing courses in a variety of languages including English, German, Spanish, French, and Italian. The most commonly offered course is the Asterisk Bootcamp, a rigorous four-day course that takes students through every major component and feature of Asterisk. Some providers offer additional courses to meet local demand. Each Training Provider meets a standard of capability and quality, and only uses training materials authorized by Digium. Every student is given a feedback questionnaire at the conclusion of the course which is then submitted to Digium for quality assurance and continuous improvement.”

Listed below is the complete Digium Partner listing. Companies interested in being a Digium partner can apply online.

  • Interoperability Partners: Polycom, Empirix, AudioCodes, AcuLab, TransNexus,snom technology, CORTELCO, CyberData Corp, Grandstream.
  • Service Partners: Simple Signal, Bandwidth.com, VoicePulse
  • Software Partners: LumenVox, Cepstral
  • Solutions Partners: Switchvox, Aspect Software, Bird’s the Word Technologies, The VoIP Connection, Critical Links, NeoPhonetics, Soft Telecom, Uniplay
  • Training Partners: amooma, WISP-e,Telappliant, ALLNET ITALIA, Australia Technology Partnerships, Avanzada 7, beroNet, Shark Distribution, AHEEVA

Asterisk Events Past and Upcoming

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Digium, true to the open-source spirit of Asterisk, is very involved with interacting with their product’s business and user community by participating in, and more recently, hosting their own conferences and trade shows. Last week, the two-day Digium Asterisk World Conference debuted at the Boston Convention and Exhibition Center. The highlight of which was of course Digium CTO Mark Spencer’s industry perspective keynote address, “The Road Ahead is Open.” and the new Switchvox Free Edition that was released in time for the conference.

This month, it is time for businesses and users in Asia to meet up and view the latest Asterisk products in display at the AsterConference Asia 2007. While the last two Asterisk-related conferences (Astricon 2007 and Digium Asterisk World) was both held in the United States the AsterConference will be held in Kuala Lumpur, Malaysia on November 19. Though not really related to Asterisk business, those who want to get some sightseeing on the side will be pleased to know the the conference venue is pretty near the famous Petronas Twin Towers. To represent Digium would be Greg Vance, Digium’s Director of Global Sales. Vance will be giving a talk about the global perspective on Asterisk. Other topics to be tackled during plenary sessions will be:

  • Quality of Voice in the SoftPBX Market: Make the Right Choice the First Time – Mr Doug Vilim
  • Why Asterisk? – Mr Alfred Chong
  • The New and Smarter Way of Communication – Dr. Daniel Ali Aman Kraehenbuehl
  • Asterisk as One Stop Solution for All The Telecom Needs – Mr. Mitul Limbani
  • A2Billing, Pre-paid and Post-paid Telecom switch and Billing Platform – Mr Areski Belaid
  • Enterprise Deployment of Asterisk as Unified Messaging System – Mr Stephen Liew
  • Speech Technology and Asterisk – Mr David Duffett
  • Security in SIP based networks – Mr Karthik Natarajan

Installing Asterisk with a GUI on Linux

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I don’t know how I missed this, but I just came across an excellent Asterisk How-to guide on howtoforge.com, the excellent site that houses an increasing amount of guides of doing many things on Linux. The article is entitled “Installing the Asterisk PBX and The Asterisk Web-Based Provisioning GUI on Linux” and it provides a good step-by-step guide on getting started with Asterisk PBX on your favorite Linux distribution.

The guide is neatly organized into chronological sessions that you can follow step by step if you want to implement your own Asterisk PBX or if you would just like to try it out.

The guide assumes a Linux Red Hat ES4 distribution though if you know your way well enough around Linux, you can easily adopt it to suit your own Linux distribution of choice.

The first part of the guide is about installing and configuring the core Asterisk package, and the guide provides clear and concise instructions you can follow. The steps looks easy enough that most Linux users familiar with fiddling with their system can easily follow it. Just in case something doesn’t work though, you’re kind of left with your own “googling” skills. At the end of the 7-step part of the installation howto, you will have a working Asterisk installation on Linux. The guide provides basic test instructions that you can use to verify if your installation of Asterisk was succesful.

Next, the how-to discusses the necessary steps needed for you to install and configure the Asterisk manager: the component of Asterisk that lets you manage your Asterisk PBX. The guide also discusses AJAM, a new feature found in Asterisk 1.4 that allows a web browser or any http-enabled application to access the Asterisk Management Interface

The second page (and the last part) of the guide is about installing the Asterisk Web-Based Provisioning GUI. Although Asterisk can be used through the command line without this component, installing the Web-based GUI will simplify and make things a lot easier if you’re maintaining an Asterisk deployment. Again, the guide provides an extensive and yet easy to follow set of instructions to guide any user trying to install the web-based Asterisk GUI.

Overall, the guide is a good starting point if you want to get your hands dirty with Asterisk PBX.

3Com Expands Portfolio with 3Com Asterisk

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3Com is actively expanding its product offerings by launching new products to cater to all businesses from small businesses to businesses with about 200 users. The three new voice products the company has added to its product listing includes two new VCX Connect models plus the 3Com Asterisk Appliance.

The 3Com Asterisk Appliance is a self-contained unit that is based on Digium’s Asterisk Appliance. It is, of course, an open source system an is designed for small businesses with only 5 to 30 phone users. Being a self-contained unit the ease of setting it up is remarkable allowing users to simply plug-and-play. 3Com Asterisk works on all of 3Com’s phone handsets. Support and service is no problem since 3Com offers full service, warranty and support for it. The list price of the 3Com Asterisk is $1,595 at present.

The release of 3Com Asterisk Appliance is a strategic decision for 3Com as they emerge from their relative non-activity in the voice space. Right now it is the company’s aim to focus on “delivering products and solutions for converged secure networks, in which voice is an application that can be readily integrated with many others.” This makes the decision to use Asterisk a sound one since, as 3Com’s senior vice president and general manager, Bob Dechant, said they chose to work with Digium because of its “position as the Asterisk leader [and] its commitment to open standards and the ease-of-use of the appliance.”

Digium, on the other hand, is still assertively marketing their product and despite all the big news surrounding it lately (Read: The awards and the acquisition of Switchvox) the Digium team continues to be enthusiastic about each new appliance that makes use of Asterisk. As Bill Miller, Digium’s vice president of product management and marketing, said, “Digium pioneered Asterisk and is now working to bring the software’s power and rich feature set to a broader array of customers. 3Com’s selection of our new appliance to offer to its own customers validates our strategy of opening new markets to Asterisk by making it user-friendly, and is also testament to the fact that Digium’s products are the gold standard in the Asterisk ecosystem today.”

Digium Acquires Switchvox

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Digium Inc., the company behind the popular open source VOIP platform Asterisk, has announced that it has acquired Switchvox, a leading provider of IP PBX phone systems for small and medium-sized businesses (SMBs). This acquisition strengthens Digium’s presence in the SMB market and provides even more avenue for the deployment and improvement of its Asterisk-based solutions.

Switchvox is the world’s largest supplier of open source IP-PBX products for businesses, with an estimated 65000 end points in operation. The company offers CPE-based solutions powered by Asterisk (with Swithcvox’s own custom code added).

To some extent, specifically in the area of making Asterisk easy to use and providing a GUI for the powerful asterisk platform that is easy to use for its customers, Switchvox embodies what Asterisk is striving to become. Switchvox has succesfully marketed to the SMB market their VOIP and hybrid solutions, building both on the power of Asterisk and their expertise on creating a user-interface that is both intuitive and user-empowering.

Switchvox also features integration with leading leading CRM applications SugarCRM and SalesForce.com, as well as with Google Maps. With its new Switchvox SMB V3.0, detailed information from SugarCRM and Salesforce.com automatically pop-up on incoming calls via the Switchboard.The integration of Google Maps makes it possible to locate where inbound calls are coming from, empowering the users more to adopt to potential leads.

Digium and Switchvox are undergoing heavy integration of the two company’s products and solutions, and plans to unveil a new product strategy and roadmap later this year focusing on asterisk-based unified communication solutions for their SMB and enterprise customers.

But until Digium rebrands and relaunches Switchvox products under the Digium brand, Switchvox products will retain their branding for the forseeable months.

With regards to proprietary components of Switchvox’s offerings, Mark Spencer of Digium expressed interest in making them open source (Digium is a big proponent of open source) to add to the current open source Asterisk.

The acquisition looks like an excellent marriage of two companies with complementary strengths in the IP PBX space, with both parties expressing interest and and excitement on the acquisition. Josh Stephen, President and CEO of Switchvox, has this to say:

“The entire Switchvox team is excited to become part of the Digium family and to be entering this new phase of the company’s life. We look forward to extending the reach of Asterisk and other Digium products, making them more accessible to more people. We have been working on a road map that we feel will change the IP PBX landscape, and with the knowledge and help of Digium we will to be able to work faster than ever to bring those features to market.”

And on their blog, Bill Miller said:

“Danny and Josh, the two respective CEOs will publish their thoughts here shortly, I am prefacing their post by telling you, “Woo hoo!!!!!!!!!” Get ready to Rock and Roll with Digium and Switchvox. When “best of breeds” get together, the results are “best of the best” so join us for the ride!”

AstLinux: An Asterisk-Optimized Linux Distribution

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Linux’s popularity and extensibility brought about by its open source design has brought many application providers to configure linux distributions that include an integrated and pre-configured versions of their applications. Also because of the nature of open source culture, it may not be necessary that the original developers of the application will be the one to “bring out” this particular distribution. Such is the case of the AstLinux Project.

AstLinux is a custom linux distribution that is built with the main purpose of running the Asterisk PBX, although the project page also says that you can easily configure other VOIP applications such as OpenSER on top of it. AstLinux is designed to address the need for embedded and commercial solutions for Asterisk, but with adept linux configuration knowledge, can be customized to suit other situations.

AstLinux can be run on multiple processor architectures and has special images for the SC100 series of single board computers. This includes machines from Soekris Engineering and PC Engines.
Because AstLinux is Linux, it can be configured to run on a wide variety of hardware platforms that the generic Linux distribution supports. Because of this, AstLinux’s generic form (or image) will run on pretty much any standard PC.

The installed AstLinux distribution occupies only 40MB, and though it has been designed to be run from Compact Flash memory, it’s perfectly fine to install it on regular hard disks. The only requirement is that it has a disk space of not less than 64MB.

Using AstLinux compared to running Asterisk on top of an un-optimized general-purpose Linux distribution gives a more efficient Asterisk PBX that requires minimal memory footprint and hardware power. Also, with the design consideration that the target machine where AstLinux will be used is a network appliance, a more efficient Linux version is included in the distribution. Eliminating unnecessary packages and components, and optimizing operations for compact flash memory which has a limited number writes and rewrites.

AstLinux looks like a good technology tool that you can include in your offered Asterisk VOIP solution.

Asterisk Bags BOSSIE Award for VOIP Telephony

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Asterisk was announced the clear winner of this year’s BOSSIE (Best of Open Source Software) Award for VOIP Telephony by InfoWorld. InfoWorld acknowledged the superiority of Asterisk above other open source VOIP technologies such as OpenPBX and FreeSwitch in their 2007 BOSSIE Special Report. According to InfoWorld,

“Asterisk is by far the most mature and scalable of the lot, and it’s taking the VoIP world by storm. Yes, it’s hideously complex in places, but also immensely configurable, and compatible with darn near everything.”

High praise indeed! But what makes it even more special is not only the fact that Asterisk won an award and much praise but that it won an award in an cateogry that is considered by many, including InfoWorld, to be “the most significant trend in networking today“.

The fact that Asterisk bagged the award is not really surprising considering the recent events that took place in the telephony world. If you recall last month we reported Genesys’ decision to support the Asterisk® open source IP PBX platform. The day it was announced was a big day for Asterisk and the entire open source world. After all Genesys’ IS the leading provider of software for major contact centres, which means that the Asterisk platform will, after the majority Genesys’s client have transitioned, be the leading VOIP telephony platform used in contact centres worldwide. With this award Asterisk scores another point for open source telephony. As Mark Spencer, CTO and founder of Digium, said,”Receiving this industry recognition from IDG and InfoWorld is not only a win for Digium, but the entire open source telephony community as a whole. Digium is committed to taking Asterisk to even greater heights and developing new ways for businesses all over the world to experience cost savings and flexibility never before possible with proprietary VoIP solutions.”

The support of Genesys’ and the award given to Asterisk though is but a reflection of th Asterisk team’s commitment to providing the best possible open source telephony engine and tool kit for FREE. It is true that Asterisk offers “unheard off flexibility in a world of proprietary communications”. It is also true that it “empowers developers and integrators to create advanced communication solutions” at the lowest price possible – zero. And it is true that Asterisk offers excellent support to its users.

Voiceroute in TMC Internet Telephony EXPO West 2007 Shows Off Druid Live

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Voiceroute showed off its Druid Live a new version of the Druid Telephony Platform during last week’s TMC ITEXPO West 2007. Voiceroute claims that with DRUID Live, DRUID is now not just the best GUI for Asterisk in the market, but is also the “first truly plug and play CDP (Cisco Discovery Protocal) based auto-provisioning of Cisco, Polycom and Linksys phones with additional VLAN based auto-provisioning of Aastra, Snom and GrandStream phones.”

During the exhibit Voiceroute showed of its upcoming Druid Live Click-to-Call VoIP hosted service with Adobe Flash for enterprises. Aside from the features that the “old” Druid telephony platform brings (as mentioned above) , according to Ming Yong, Voiceroute CEO, Druid Live allows “enterprises to embed interactive flash widgets into their flash video, rich media ads, text ads and web sites to allow interested customers to seamlessly connect using voice and text with a sales or customer support representative without installing any java soft-phone client on the customer side or server side software on the SME side.”

Druid Live’s beta version is set for release this October. In the meantime, thos einterested in DRUID will be happy to know that their current version is packed with excellent features that makes a VOIP telephony platform relatively easy to install and manage.

Features of the current DRUID version include:

  • Extension Management Wizard – allows easy removal and addition of phones and users
  • Trunks Management Wizard - allows instant configuration of inbound / outbound trunks
  • Dialplan Wizard - allows the creation of Asterisk dialplan controls without needing to learn about nor understand the complex dialplan languages needed to be able to route calls into your system. The Dialplan Wizard is so easy to use that even a first time Asterisk user can figure it out.
  • Hardware Wizard (E1/T1/PSTN/TDMoE) – allows easy configuration of an Asterisk server with Digium / Sangoma E1, T1, FXO, FXS card as well as GSM gateways and Channel Banks.
  • Druid User Portal – allows you Asterisk/Druid users to check their voicemail and configure other call options such as Simultaneous-Ring and Follow-Me features.
  • Live Java Console – a remote Asterisk call control utility that allows operators/administrators to see call activity in real-time and hang-up, transfer or originate a call remotely.