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October 31, 2007

Asterisk PBX Takes Hold in the Enterprise

PIKA Technologies Inc., a developer of media-processing hardware and software, today announced the results of an analysis of the usage habits of over 300 Asterisk developers across the globe. The announcement coincides with the start of Digium Asterisk World at this year’s Fall VON conference. From the results, it is clear that the Asterisk platform is becoming much more prevalent in the enterprise.

"Some of the most compelling data that we collected was simply around the wide variety of applications that Asterisk is now being used to address," said Terry Atwood, vice president of sales, marketing and customer care at PIKA. "Of course, the early adopters are still there, but from a purpose and density standpoint there is now a much broader range of applications being built on Asterisk, ranging from individual companies seeking to provide their own telecoms services, to application development firms that are using Asterisk as a platform to develop systems for sale to the business and government markets throughout the world."

Common applications being built on Asterisk included normal office telephone systems, IVR/self-service systems, systems for use in inbound and outbound call centers, and hosted telephone and VoIP long distance systems. Other unique applications included distance-learning systems, fax blasting platforms, video platforms and gateways. The study also showed that educational institutions around the world are using Asterisk in their laboratory environments, allowing students to get hands-on development experience in a real-world scenario.

Results also show that the main Linux distributions being used by Asterisk developers in the study are CentOS (34%), Debian 4 (20%) and Fedora (16%). Ubuntu and Slackware are both being used by about 6% of developers surveyed, with a host of other platforms being used by less than 1% each.

The version of Asterisk being used was split between 1.2 (49%) and 1.4 (46%), with the final 5% still using 1.0. Additionally, Asterisk was observed to have spread beyond the confines of North America, with a third of those in the study hailing from Europe.

The analysis was conducted as a result of a free board giveaway program, run by PIKA during July and August 2007. The program allowed Asterisk developers to order one of PIKA’s analog or digital Asterisk boards free of charge. Signing up for a board required imparting some detailed information about the user’s/developer’s Asterisk environment. Based on this information, PIKA was able to present its conclusions about current Asterisk usage. The full data from the program can be viewed at http://www.pikatechnologies.com/products/PIKAforAsterisk-market_study.htm.

PIKA is exhibiting and presenting at Digium/Asterisk World as part of Fall VON, October 29 through November 1 at the Boston Convention and Exhibition Center in Boston, MA, and will be holding a draw for 2 FREE appliances. Interested parties can visit and learn more about the company's new Appliance and Gateway products at Booth D1325, part of the Asterisk World Pavilion.

At Fall VON, PIKA business development manager David Clarke will participate in the panel discussion, "Echo Cancellation: Even Packets get Voice Quality Issues" taking place from 3:45pm to 5:00pm on Thursday, November 1. The panel will discuss why packet-based networks are plagued with problems like echo cancellation, phase and jitter and the various approaches to rectifying these issues.

Source: Danny Sullivan w/ inMedia 

 

October 30, 2007

Cisco Enters the WiMAX Market

One week after the International Telecommunications Union approved WiMAX as a 3G technology, Cisco announced it is moving into the WiMAX space with the purchase of Navini Networks, a major manufacturer of WiMAX networking equipment.
At the time of the ITU's approval, the WiMAX Forum predicted the move "significantly escalates opportunities for global deployment, especially within the 2.5-2.69 GHz band, to deliver mobile internet to satisfy both rural and urban market demand."

Evidently, Cisco thinks so, too. The deal will cost Cisco $330 million and is expected to be complete in the first half of 2008.
 
Source: Personal Bee 

October 29, 2007

Mark Spencer: "Consider open source VoIP, think twice about Hybrid-Hosted"

Companies selecting a VoIP solution must choose from a dizzying array of options, including whether a hosted, hybrid-hosted or premise-based telephony system will work better for them, and whether the benefits of open source outweigh the potential risks.
 
Digium CTO and Asterisk creator Mark Spencer called the proprietary hybrid-hosted model "very evil" during the Internet Telephony Expo West 2007 in Los Angeles, leading to a conversation-starting blog posting by SearchNetworking.com site editor Amy Kucharik. Kucharik recently spoke with Spencer to get some clarification on his position and controversial comments, and to hear his thoughts about open source VoIP and what it means for business users.
 

Why is proprietary hybrid-hosted VoIP 'very evil'?


Mark Spencer: A lot of this is a bit of a hyperbole in terms of the scale I try to put things in. Are the lines quite so black and white? But the biggest problem with hybrid-hosted as it is currently implemented is that it actually takes away even more choice from the customer than a traditional proprietary product. The reason for that is that under a hybrid-hosted model, the ability to control the system is removed from the customer and placed in the hands of the vendor. So when you think about it, open source is trying to move things to where there is customer control; the hybrid-hosted model is that you really don't even have access to your configuration information because it's all held there. If they change what the feature set is or how it's presented, you don't really have any choice about how to use that. It's not so much that the hosted model itself is inherently bad; there's certainly a lot of technical benefit to having a hosted environment. It's an easy way to get started. But it is, I think, dangerous to customers not to be able to get out of it if they outgrow the sise that makes sense or if they become unhappy with that vendor. They're really locked in even more than they are with a traditional proprietary vendor.

Do you think there are certain companies for which it is more or less dangerous? The hybrid-hosted model seems most beneficial for SMBs, which don't have the staff or the desire to manage the phone system in-house; they just want to pay somebody and have it work.


Spencer: I think that it's one of several models that can work for an SMB. Some SMBs like to have premise-based solutions if they're cost-effective enough, and some SMBs want to have something that's hosted; I mean, it just kind of depends on the company. But when your SMB starts to turn into more of an enterprise, as you start to grow, that's when this starts to be an acute problem because you're really locked in to this vendor in a way that's very difficult to break away from.

And not only that, but what you see in the product today may or may not be what the product is tomorrow. Think about how many people have used Windows XP, and now you have Windows Vista that's come out. Imagine if all of a sudden Microsoft changed it so that when you went to your computer it was now running Vista because they thought that was the better system. Think about how many people really don't like Vista right now. If your operating system were hosted, you really wouldn't have a choice. Whatever the person hosting it decided you should have is what you'd have. It just doesn't feel like a safe model for someone who's used to the open source world, where you have a tremendous amount of control.

Click Here to Continue Reading

 

October 24, 2007

Digium Unveils New Corporate Headquarters in Huntsville

 
Digium, Inc., announced the grand opening of a new corporate headquarters to support the company's rapidly growing open source telephony business. A ribbon-cutting ceremony at the new facility, located at 445 Jan Davis Drive NW, is scheduled for October 23. Digium, which has called Huntsville home since its founding in 1999, is the leading worldwide provider of open source VoIP solutions.
"Digium is experiencing exciting growth as more and more businesses realize the technical superiority, flexibility and low cost of open source VoIP solutions," said Danny Windham, president and CEO of Digium. "We are extremely thankful to the Huntsville community for contributing to our current and future success and look forward to even greater things as we occupy our new state-of-the-art facility."

The Digium ribbon-cutting ceremony will begin at 10:30 a.m. with Chamber of Commerce Officer Evan Quinlivan serving as master of ceremonies. The ceremony will include speeches from Huntsville Mayor Loretta Spencer, Digium Founder and CTO Mark Spencer, Digium President and CEO Danny Windham and Clay Smith, son of the late ADTRAN CEO and Digium board member, Mark C. Smith. Lunch and tours of the new facility will be offered immediately following the ceremony.

Digium's new facility will also feature Huntsville's Aroma coffee shop which will be located on the first floor and will serve freshly brewed coffee and pastries daily from 7 a.m. to 4 p.m.
 
Source: PR Inside
 

October 19, 2007

Nerd Vittles Leaves Trixbox and Annouces PBX-in-a-Flash

 
 
Note:  I have to agree about the direction of Trixbox cough cough Asterisk cough Platform.  It seems like they have lost touch with there roots.  Just because you purchase an open-source project does not mean you have some ownership of the underling code.  It belongs to the community as a whole and if anyone would have any rightful claim it would be our boy Marc Spencer.  So there, I said.  I have not really wanted to comment on this because A. there are some conflicting interests with my consulting services on the Asterisk iPBX and B. It is really a waste of my users and my own time.  But with our respected Ward Mundy making that comment I felt I need to be honest about my feelings on this subject.
 
We’ve been keeping a low profile for a few weeks, and now it’s time to let the cat out of the bag. As some of you know, we just haven’t been thrilled with the direction of the trixbox project lately. Without boring everyone with a lot of detail, suffice it to say that it’s just gotten a little too proprietary, too closed, and too commercial for our open source, puritanical tastes.

So today, with a bunch of help from some really sharp folks, we embark upon a new open source project that we hope will become the best-of-breed Asterisk-based development platform. Our design goals are simple: a very modular system that meets the needs of Asterisk experimenters as well as those looking for a reliable, scalable, IP-based business telephony solution with all the bells and whistles.

Our up front promise is to keep the project open, participatory, reliable, and fun. After all, that’s what the Asterisk revolution was and is all about. The plan is to provide a free ISO-based offering for home or office use that will run on a dedicated Linux machine. There also will be a VMware image that will run on a Windows desktop. And, for the Mac desktop, we’ll provide both a VMware and a Parallels image.

Click Here to Continue Reading 

 

 

WiMAX Becomes Recognized 3G Technology

On October 18, 2007 the ITU Radiocommunication Assembly in Geneva adopted inclusion of WiMAX into the IMT-2000 family of technologies. This significant decision will make WiMAX as one of the approved IMT technologies and increase the adoption WiMAX in the world.  This latest decision puts WiMAX on the same level playing field as other 3G technologies such as UMTS”.

TEMA Vice President and Motorola’s Regional Director of Government Relations Bharat Bhatia who is participating in the ITU conference at Geneva and played an active role in the conference said: “Today ITU took an important step toward ensuring that the public will benefit from the most advanced wireless technologies when it accepted WiMAX into the IMT-2000 family of technologies.

Inclusion of WiMAX into the IMT-2000 frequency bands is especially important for India as we work to expand the broadband penetration which will bridge the countries digital divide and tremendously benefit our consumers”.

N K Goyal, Chairman Emeritus TEMA commenting on the decision said “TEMA applauds today’s decision by the ITU Radiocommunication Assembly to approve WiMAX as part of the IMT-2000 family of standards. India’s entire telecom industry appreciates the efforts made by the Indian delegation at the ITU conference in playing a key leadership role in this decision.

Source: India PR 

October 17, 2007

Asterisk 1.6 Release Management Proposal

 
 
Greetings,

A few weeks ago, I proposed to this list that we create a new release series that is managed with a short release cycle to introduce smaller sets of new features.  I also wanted to increase the emphasis that we put on testing new sets of functionality for potential regressions.

The feedback on this list was positive, as was all of the feedback I have received directly.  I spoke to people about this a lot at Astricon, and received no negative feedback.
So, I would like to move ahead with formalizing this new release series, Asterisk 1.6.  I have documented the new release policy that will apply to this release series, as well as some of the history that inspired these changes to release management.

I have included the document and would appreciate any feedback from the development community.

In short, as long as there are no significant disagreements, I plan to create an Asterisk 1.6.0-beta1 snapshot next week.  I will continue to make beta tarballs until we are comfortable moving to release candidate status.  I will then create an Asterisk 1.6.0 branch and start making release candidate snapshots, named 1.6.0-rc1, etc.  After a reasonable amount of testing and no known regressions introduced by the new changes in 1.6, I will release 1.6.0.  The 1.6 release series will continue to be managed in the manner described in the included document.

The current levels of maintenance of Asterisk 1.2 and 1.4 will not change at any point in the near future.

Let me know if you have any questions, comments, or concerns.

Thanks,


---------------------------------------------------------------------------

             Asterisk 1.6 Release Management

                     Russell Bryant
                      Digium, Inc.
                    October 17, 2007

Contents

1 Introduction                                                                1
2 Current Problems                                                            2
    2.1 CVS Head, 1.0, 1.2, 1.4 ... . . . . . . . . . . . . . . . . . . . .   2
    2.2 Shot in the Foot . . . . . . . . . . . . . . . . . . . . . . . . . .  2
3 Changes for Asterisk 1.6                                                    2
    3.1 SVN Branch Layout . . . . . .       . . . . . . . . . . . . . . . . . 3
          3.1.1 team branches . . . . . .   . . . . . . . . . . . . . . . . . 4
          3.1.2 trunk . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
          3.1.3 tags . . . . . . . . . . .  . . . . . . . . . . . . . . . . . 4
          3.1.4 branches/1.6.X . . . . .    . . . . . . . . . . . . . . . . . 4
    3.2 SVN Commit Workflow . . . . .       . . . . . . . . . . . . . . . . . 5
          3.2.1 Small new feature . . . .   . . . . . . . . . . . . . . . . . 5
          3.2.2 Significant New Feature     . . . . . . . . . . . . . . . . . 5
          3.2.3 Small Bug Fix . . . . . .   . . . . . . . . . . . . . . . . . 5
          3.2.4 Invasive Bug Fix . . . .    . . . . . . . . . . . . . . . . . 5
          3.2.5 Security Fix . . . . . . .  . . . . . . . . . . . . . . . . . 6

1      Introduction

This document describes the release management strategy for Asterisk 1.6. It is vastly different than what has been done for previous releases, so having a clear understanding and agreement on how it will be done is critical.

                                       1


2     Current Problems

Before diving into how Asterisk 1.6 will be managed, it is worth reviewing the release history that has inspired these changes.

2.1     CVS Head, 1.0, 1.2, 1.4 ...

Just a few years ago, everyone that was using Asterisk obtained it by checking out the latest code from the development tree, which was CVS Head. This meant that Asterisk was a very fast moving target, and users just hoped that they caught it on a good day.
   
    In the Fall of 2004, Asterisk 1.0 was released. This began the era of managed releases of Asterisk. The policies established at this point carried on into 1.2, and then 1.4. The simple policy for release branches was ”only bug fixes”. The idea is that if you keep code changes to an absolute minimum, then there is a smaller chance of new bugs getting introduced.
    Well, great news! The release management for the past few years has been a success, to some degree. Releases have proven to get very stable and the entire Asterisk user community has transitioned to using the releases.

2.2     Shot in the Foot

    The problem with this release management comes up when we want to release a new version. Over a few years, we have gone from everyone running the latest development code, to only a core set of developers using it. So, bugs aren’t being discovered during development, and only show up when it gets into a release.
   
    When the release is made, we are stuck tracking down weird bugs intro- duced during a year to a year and a half’s worth of development. This is not fun for anyone. This is the exact reason it has taken so long for Asterisk 1.4 to really get stable.

3     Changes for Asterisk 1.6

Asterisk 1.6 introduces a new release management style for the Asterisk project.

                                       2


    The 1.6 version will receive new functionality in smaller increments. In- stead of doing doing another year of development before releasing an ex- tremely large set of changes as Asterisk 1.8, we will be adding things into each release of 1.6.
   
While trunk continues to receive new features and architectural improve- ments, we will make 1.6.X release branches every month or two. After the branch is made, we will make release candidates available while the devel- opment team and community members test for regressions introduced by what new things have been introduced in this release. The timeframe stated here is intentionally vague because releases will be determined based on code quality, and not an arbitrary date.
   
Meanwhile, while a 1.6.X release branch is being tested, new things can continue to be merged into trunk. After an official 1.6.X release is made, a new 1.6.X release branch will get created and the process will start over.

3.1     SVN Branch Layout

    • asterisk/trunk
    • asterisk/team
          – asterisk/team/russell
          – asterisk/team/kpfleming
          – asterisk/team/file
              ∗ asterisk/team/file/cool-branch1
              ∗ asterisk/team/file/cool-branch2
    • asterisk/tags
          – asterisk/tags/1.4.11
          – asterisk/tags/1.4.12
          – asterisk/tags/1.6.0-rc1
          – asterisk/tags/1.6.0-rc2
          – asterisk/tags/1.6.0
          – asterisk/tags/1.6.0.1
          – asterisk/tags/1.6.0.2

                                       3


    • asterisk/branches
         – asterisk/branches/1.2
         – asterisk/branches/1.4
         – asterisk/branches/1.6.0
         – asterisk/branches/1.6.1
         – asterisk/branches/1.6.2

3.1.1    team branches

For anyone with commit access, this is where large all new developments should go first. This includes any significant new features or invasive bug fixes that need extra testing. The changes should not move into release branches until they are reasonably tested and considered ready for release.

3.1.2    trunk

This is where all of the new developments go once they are considered ready for release. 1.6 sub-releases will be branched off of trunk every month or two. Special care must be taken by those with commit access to not introduce a large number of extremely invasive changes to the same parts of the code in the same 1.6 release cycle, as we want to make tracking down regressions from invasive changes as easy as possible.

3.1.3    tags

Tags are simply release snapshots. A tag is made every time that a tarball is created and reflects exactly what was in the release.

3.1.4    branches/1.6.X

1.6 branches will be created every month or two. They will include all of the new functionality committed to trunk since the previous 1.6.X branch. After the branch is created, release candidates will be previously tagged off of the branch while testing is being done. After a reasonable amount of testing has been done and the development team is comfortable that none of the new things introduced in the release have caused any regressions to the best of their knowledge, then the official 1.6.X release can be made.

                                        4


    If any significant regressions are found after the 1.6.X release, then com- mits can be made to the 1.6.X branch to fix the issue, and 1.6.X.X releases can be made.

3.2     SVN Commit Workflow

3.2.1    Small new feature

Commit directly to trunk.

3.2.2    Significant New Feature

Commit to a team branch. Once the feature has received a reasonable amount of testing, then it can be committed to trunk, as long as it does not conflict with other significant changes made to the same code in the current 1.6 release cycle. If that is the case, then the commit of this significant change should wait until the next release cycle.
    Contact Russell Bryant or Kevin Fleming for help in deciding, if necessary.

3.2.3    Small Bug Fix

   1. Commit to the 1.4 branch
   2. Commit to the current 1.6.X branch that is in testing, but only if the
       bug is a regression introduced in that specific 1.6.X release. If it is a
       bug that has been around longer than that, it will have to wait until
       the next 1.6.X release.
   3. Commit to trunk

3.2.4    Invasive Bug Fix

Commit to a team branch. Once the patch has received a reasonable amount of testing, then it can be committed to the 1.4 branch and trunk, as long as it does not conflict with other significant changes made to the same code in the current 1.6 release cycle. If that is the case, then the commit of this significant change should wait until the next release cycle. Also, if the fix is for an issue that was introduced in the current 1.6.X branch that is in testing, then the fix may be put there.
    Contact Russell Bryant or Kevin Fleming for help in deciding, if necessary.

                                        5


3.2.5   Security Fix

   1. Commit to the 1.2 branch
   2. Commit to the 1.4 branch
   3. Commit to the current 1.6.X branch that is in testing, as well as the
      past three 1.6.X release branches so that sub releases of those can be
      made that include the fix.
        • Note that the number three here is arbitrary. It may change based
           on what community members would like to see.
   4. Commit to trunk.

                                        6


--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

Polycom and 3Com Partner to Deliver Enterprise VoIP Solution

Polycom and 3Com announced a co-branding agreement for 3Com to OEM and jointly market Polycom's wired Soundpoint IP and conference phones. The companies also intend to include future integration in areas such as voice over wireless LAN (VoWLAN), mobility and integration of other popular business applications such as CRM and VoIP security.

Today's announcement augments Polycom's distribution channels to include 3Com's network of resellers and distributors, including a worldwide community of service providers who offer both IP trunking and hosted VoIP services. Today's news also adds to 3Com's portfolio Polycom's HD Voice lineup of phones, which enrich the customer communications experience with high definition voice quality that quadruples that of traditional landline phones.

"The combination of 3Com's strong distribution network with Polycom's voice technology expertise makes this the ideal solution for our customers," said Sunil Bhalla, senior vice president and general manager of Voice Communications for Polycom. "Our voice solutions will bring the richness of HD Voice quality to 3Com customers, affording them crystal clear communication that greatly improves productivity.


We also share a vision with 3Com of providing open, non-proprietary solutions to our customers. And with the addition of 3Com as a strategic distribution and technology partner, we look forward to deepening our relationship by expanding our efforts into key areas such as wireless and business productivity applications."

Polycom phones will be seamlessly integrated with all 3Com VoIP platforms, including the newly announced 3Com(R) VCX(TM) Connect 100, VCX Connect 200 and 3Com Asterisk Appliance, as well as its existing 3Com VCX Enterprise system, including a 3Com IP Telephony Solution for IBM System i. 3Com's family of IP Telephony servers are used by companies of all sizes worldwide.

"The addition of Polycom's voice solutions to our portfolio of 3Com IP telephony solutions and handsets will enable us to offer customers a wide range of choices to meet their unique communications needs," said Bipin Mistry, vice president of 3Com's voice business. "Coming on the heels of our introduction of the new 3Com VCX Connect platforms and the 3Com Asterisk Appliance, this partnership means that organizations with as few as five employees will enjoy the same robust VoIP offering as those with thousands of employees.

The co-branded 3Com/Polycom phones will be generally available for purchase through 3Com North American channels beginning in November 2007 and are expected to be available on a worldwide basis no later than Jan. 1, 2008.

Source: PR Newswire 

October 15, 2007

Polycom Fights Cisco With New Telepresence Video System

Polycom is going head-to-head against Cisco Systems with the launch Monday of its latest telepresence high-definition videoconferencing system. Polycom's new Telepresence Experience High Definition (TPX HD) 306M system is a full room videoconferencing setup that aims to provide video calls that seem lifelike, closely approximating the dimensions and detail of face-to-face meetings. It is priced one-third lower than the comparable offering from Cisco.

The TPX is the latest addition to Polycom's line of telepresence offerings, which launched in May 2006 with the rollout of its RealPresence Experience (RPX) portfolio.

While the RPX provides what Polycom calls "immersive telepresence" with a complete standardized full-room structure with rear-projection displays to accommodate up to 28 users per room that even includes the ceiling overhead, the TPX is a smaller setup built for six users per room.

The TPX includes a wall unit that incorporates three 60-inch HD plasma displays, videoconferencing equipment, cameras and a high-fidelity stereo system. The setup also includes a conference table, 360-degree ceiling microphones and a touch-screen control system.

Click Here to Continue Reading 

 

October 11, 2007

Criminals using Skype to Talk Securely

Criminals are taking advantage of Skype's 256-bit encryption and the internet's unpredictable routing scheme to set up drug deals and other crimes, say Swiss law enforcement officials. According to swissinfo, a Swiss Broadcasting Corporation company, local police are investigating dozens of drug-related cases where Skype has been the medium of communication between gang members and client.
"Criminals know that the police have difficulties monitoring Skype," it quotes a Zurich prosecutor of drugs and organised crime, Christoph Winkler, as saying.

According to Bernhard Weder, head of a Swiss federal working group that is researching how to monitor internet phone calls, Skype is "extremely cleverly built", it said.

Weder believes Skype can decode calls, but said the firm has not been co-operative, a charge the company's PR agency denied.

Source: Computer Weekly 

October 10, 2007

Open Source-based Telephony Solutions Boost the IP Telephony Market

 
 
The open-source telephony market is still in the early-adopter phase. While it took years for Linux to achieve mainstream status, Frost & Sullivan believes that the path to the commercial acceptance of open-source telephony will be much faster. The Linux movement has paved the way and has taken care of initial reservations about open-source technologies as a business-grade option.
 
New analysis from Frost & Sullivan Open-Source Telephony Solutions, finds that the installed base for business-grade, licensed, open-source telephony in terms of telephony lines/users is a little more than 200,000 users in North America.
If you are interested in a virtual brochure, which provides manufacturers, end users, and other industry participants with an overview of the latest analysis of the Open-Source Telephony Solutions, then send an e-mail to Mireya Castilla, Corporate Communications, at mireya.castilla[at]frost.com with your full name, company name, title, telephone number, city, state, country and e-mail address. Upon receipt of the above information, an overview will be sent to you by e-mail.
 
"One of the most common factors holding back the adoption of open-source telephony is the assumption that the total cost of ownership is much higher than in proprietary cases," notes Frost & Sullivan Industry Analyst Krithi Rao. "While this may have been true with ‘black-box’ solutions in the time division multiplexing (TDM) market, Internet Protocol (IP) telephony changes the paradigm."
 
Deploying IP telephony frequently involves network infrastructure upgrades, spending on licensing fees for various applications that can be deployed to take advantage of the IP infrastructure and implementing redundancy that is needed because it is IP. Accordingly, only the customer’s deployment scenario will determine whether a proprietary IP solution will be more economical than an open-source software-based solution.
 
There seem to be multiple deployments of open-source telephony software or instances where such software has been incorporated into commercial products, which have not been brought to light. In many cases, since the free version of the software is downloaded and deployed, vendors supporting open-source software are not aware of the deployment and not able to capitalize on success stories.
 
Until open-source telephony gains recognition as a business-grade option, these customer cases that are possibly very successful are not likely to receive extensive publicity. "The perception of open-source being risky has prevented users and proponents from being completely vocal about actual implementations," explains Rao. "The common motto in the industry is: You never get fired for deploying Cisco, Avaya, or Nortel."
 
Successful open-source telephony deployments by value-added resellers (VARs) and partners, which are typical channels for the closed-source IP telephony market, will help validate the reliability of open-source solutions. For customers, choosing a partner that is experienced with IP telephony deployments and has certifications such as digium certified asterisk providers (DCAP) should ensure a successful business-grade deployment.
 
Source: News Wire Today 
 

Fake caller ID: Fun, Legal and Easy

 

 

Note:  He is right, you can't trust caller ID these days.  I do know that is someone sends no caller ID then I don't answer.  It's usually a telemarketer or some congressmen trying to give me an award for donating money to there party or associations.  In our current environment I am not sure how long that will last.  If you are trying to get in touch with your local press try changing it to some major organization's ID and see if your call gets picked up faster?  I bet it would.

He cited some legislation that was put through in 2007 and is currently stuck in the Senate.  "The law would outlaw causing "any caller identification service to transmit misleading or inaccurate caller identification information" via "any telecommunications service or IP-enabled voice service." Law enforcement is exempt from the rule."

Here is an excerpt from the article: 

 

"Caller ID information is not to be trusted. Judging by the reactions I've gotten from colleagues and friends recently after they've been the victims of spoofed-ID demonstrations, it's not common knowledge that caller ID information, primarily the phone number that often appears on the recipient's telephone display, can be easily faked. Best of all for the mysterious caller, it's not illegal in the U.S. (except in cases where fraud occurs). Calls for the purpose of amusement or revenge are perfectly legal."

Click Here to Continue Reading

Asterisk Version 1.4.13 Released

 
 
The Asterisk Development Team has released version 1.4.13
 
This release fixes a couple of security issues in the implementation of IMAP storage for voicemail.  One of the issues is remotely exploitable.  Any systems that do not use IMAP storage for voicemail are not affected by these issues. For more details on this issue, see the Asterisk security advisory here:

 * http://downloads.digium.com/pub/asa/AST-2007-022.pdf

This release also contains some other bug fixes that have been merged in the past week or so.  The other fixes include resolutions for a few different deadlocks, a couple of problems in res_jabber, chan_sip and RTP fixes, and a few more minor issues.  See the ChangeLog for a full listing of the changes:

* http://downloads.digium.com/pub/telephony/asterisk/ChangeLog-1.4.13

Thank you very much for your support!

October 09, 2007

3Com to Resell Digium's Asterisk Appliance

Note:  More corporate traction the better.  Asterisk needs to be brought into the mainstream light so they become a brand associated with business phone systems in conversation.   Where is my Polycom branded unit?

"
Huntsville-based Digium Inc. said Monday that it has formed a partnership with networking giant 3Com Corp. Under terms of the agreement, 3Com will sell Digium's Asterisk Appliance to small businesses that need voice-over-Internet Protocol telephone service.

Digium is the creator of Asterisk open-source software for VoIP. The company's Asterisk Appliance is its first hardware device with imbedded Asterisk software for private-branch exchange, or PBX, phone systems. The appliance supports both VoIP and analog systems, allowing users to keep their current equipment.

"3Com is focused on delivering products and solutions for converged secure networks, in which voice is an application that can be readily integrated with many others," Bob Dechant, 3Com senior vice president and general manager, said in a news release. "We chose to partner with Digium because of the company's position as the Asterisk leader, its commitment to open standards and the ease of use of the appliance."

3Com, based in Marlborough, Mass., is a global company that provides secure and converged networking services that enable customers to manage voice, video and data in a secure network environment. 3Com last week announced a deal to be acquired by private-equity firm Bain Capital Partners for $2.2 billion.

"Digium pioneered Asterisk and is now working to bring the software's power and rich feature set to a broader array of customers," Bill Miller, Digium vice president of product management and marketing, said in a news release. "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."

Source: Huntsville Times 

Jajah and eBay in VoIP Access Showdown

VS.
Internet voice service provider Jajah on Thursday said that web auctioneer eBay blocked the use of the startup's embedded voice links on its auction site, a move Jajah said raises concerns about anti-competitive behavior.

Jajah said eBay removed the listings of sellers who used Jajah’s technology, which enables potential buyers to connect with sellers through a click-to-call button embedded in their ad.

eBay sent emails to its sellers in the U.S. and Italy instructing them that Jajah’s technology, called Buttons, violated eBay’s link policy.

eBay’s policy seeks to protect its site from unauthorized or insecure links that could harm its business, but Jajah charges that eBay’s ownership of Skype, a VoIP firm, makes the auction giant’s argument specious.

“The fact that eBay acquired Skype for billions of dollars showed that they thought that VoIP would be a value-add for its community,” said Roman Scharf, co-founder of Mountain View, California-based Jajah. “Now that we are offering VoIP to the eBay community, eBay finds it to be a problem.”

According to Mr. Scharf, eBay, which acquired Skype in October 2005 for $2.5 billion, is pursuing the no-Buttons policy in the U.S. and Italy but not in other markets such as Germany and the U.K.

San Jose, California-based eBay said Jajah’s Buttons violates its links policy in every country in which eBay does business, and that it will continue to remove listings that include Jajah’s Buttons.

“We don’t allow Skype buttons in our listing pages either,” said Catherine England, an eBay spokeswoman.

“There are Skype links in specific areas and we have started testing Skype buttons but anytime a link directs users off of our site, it creates opportunities for fraudsters,” Ms. England said. “The policy is about safety for our members.”

Jajah’s Buttons, introduced on Monday, allow sellers for instance to place widgets in blogs or emails and have potential buyers make an inexpensive calls to sellers without the seller giving out his or her phone number.

“eBay’s policy makes reference to insecure or political content and the like, and our Buttons are nothing like that,” Mr. Scharf said. “We don’t take the user to a rival or fraudulent site or any such thing. Using Buttons just means that the user’s phone rings.”

Mr. Scharf said he is willing to offer eBay a commercial olive branch – a piece of the action.

“Every single Jajah call means money in the pocket of our partners,” he said. “eBay has its choice. It can simply tolerate Jajah or support Jajah but either way it will mean revenue for them.”

With VoIP application providers increasingly embedding voice communications in emails, blogs and web sites, it seemed inevitable that some Internet business would eventually move to block non-sanctioned VoIP services.

But the fact that the web business is eBay, which owns Skype, the best known VoIP provider in the world, brings up issues of open markets and competition that could draw regulatory attention.

VoIP sits on a precarious boundary between the Internet and telecommunications. On the one hand, the Internet is cheered as the ultimate open marketplace. On the other, telecommunications is one of the most heavily regulated industries in the world.

“Once voice becomes an application embedded in a web page, the notion of utility telephony begins to crumble,” said Will Stofega, an analyst with IDC. “Of course, if there is a compelling advantage or difference between clients, that may change the game and an end user may demand their favorite client.”

If eBay users challenged the auction site’s policy, then the issue could get on the U.S. Federal Communications Commission radar, said Mr. Stofega.

Source: Red Herring

Marc Spencer & Asterisk PBX Moment

 
 
Note:  Okay, I hope you enjoyed my obligatory Marc Spencer picture.  I was reading Tristan Rhodes post about the current Asterisk product lineup and some of the recent developments.  I think Asterisk is heading in the right direction. 
 
SwitchVox is a good step to compete directly with Shoretel and Trixtangle hehe.  I hope they keep decent margins in the formula to take care of there fiercely loyal Asterisk resellers and consultants.  The best thing that can happen is to have the margins left for the resellers so that keep a large volume of competent Asterisk installers that can support the product on a large scale.
 
I like where the Lumenvox's speech reconition product is going also.  Interactive voice application is already big and is only going to get bigger. 

Excerpt from his article:

 "

Digium is the company behind Asterisk, the popular open source PBX. Digium was founded in 1999 by Mark Spencer, the creator of Asterisk. Since then, Asterisk has been deployed around the world on millions on computers. Despite that fact, Asterisk still does not have a large market share of the PBX market. Why is this? In the past, there were many reasons for this:

  • No brand recognition of Asterisk
  • No proven track record of successful implementations
  • Commercial support was needed
  • No Linux expertise on staff
  • Afraid to use open source software

These concerns were once valid, but today most of them have been addressed by Digium.

"

Click Here Continue Reading
 

Skype's Niklas Admits the Obvious "Ebay Overpaid"

 
 
Note:  I will admit it seemed a tad high but they do have a huge installed user base.  I think they should spend some time working on more creative ways to monetize their property without spamming it with ads.  (hint hint, contact me).
 
Skype co-founder Niklas Zennstrom admitted today that eBay's original valuation of Skype was too high, reports Reuters. It may not be a surprise--pundits have been saying as much all last week--but candid admissions are always nice to hear. His comments came at a Hungarian technology conference.
Ebay trimmed as much as $1.2 billion off the $4.3 billion purchase price after certain earn out targets were not met.

Zennstrom added that he believes Skype is growing at a satisfactory pace and that, "You need to look at the long-term value of companies." Alas public companies don't often have that luxury--shareholders want results on a quarterly basis. Zennstrom's other startup, online video purveyor Joost, opened to the public last week.

Source: Wired.com 

October 02, 2007

Asterisk 1.4.12 and Asterisk-addons 1.4.3 Released

The Asterisk Development Team has announced the releases of Asterisk 1.4.12 and Asterisk-addons 1.4.3.

The Asterisk-addons release contains just a few fixes for the modules in that package, but the Asterisk release contains a large number of bug fixes for all parts of Asterisk.

There are many areas that have been significantly improved by various fixes.  Those include the IAX2 channel driver, Queues, timezone handling, AEL, the Manager Interface, MeetMe, AGI, the SIP channel driver, Music on Hold, Jabber, the Gtalk channel driver, and more.

The listing of all changes made in these releases can be seen in the ChangeLog:

http://svn.digium.com/view/asterisk/tags/1.4.12/ChangeLog?view=markup
http://svn.digium.com/view/asterisk-addons/tags/1.4.3/ChangeLog?view=markup

The releases are immediately available for download from:
http://downloads.digium.com/pub/telephony/asterisk/

Thank you very much for your support!

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