Digium Mobile App For SwitchVox Wins 2011 Enterprise App Award from CRN

Digium Mobile App For SwitchVox http://snapvoip.blogspot.com/
Digiums SwitchVox has won the prestigious "The Appys" award, 2011 Enterprise App Award from CRN.

"The mobile workforce demands the same communications experience whether on the move or at the office. Switchvox offers fixed mobile convergence and complete mobile integration at no additional cost to our business customers. It's gratifying to receive this honor from CRN which further illustrates the tremendous value that Digium brings to its resellers and customers." Leslie Conway, vice president of global marketing for Digium,stated.

Switchvox Mobile applications run on the iPhone, Blackberry, Android phones and tablets.The users of the Switchvox Mobile applications, can make, receive, transfer and record calls as if they were calling from their office extension. They will also be able to check messages, change call rules, modify recorded greetings, access a company directory, review call history and more.
Via Digiun Press Release.

Asterisk 10, Asterisk 2.0, And The Evolution Of Asterisk!

Asterisk 10, Asterisk 2.0 http://snapvoip.blogspot.com/
If you thought I am missing 1. from Asterisk 10 in the title, this time you are wrong (I am prone to typos and I do not check my posts for faults very often).
Digium's Director of Software Engineering, Kevin Fleming has posted a lengthy article explaining the transition of Asterisk 1.10 to Asterisk 10.

"The conclusion that we’ve reached, and that we hope you’ll agree with, is that Asterisk is always going to be Asterisk, and that you don’t need a "1." prefix on the version number to be able to identify it. So, starting with the next major release, we’re going to drop the "1." completely. The next major release, which was going to be Asterisk 1.10, will now be just "Asterisk 10" and subsequent major releases will be "Asterisk 11", "Asterisk 12", and so forth.

We’ll continue with our plan to have both standard and long-term support releases of Asterisk, and we’ll update the Asterisk Project Wiki with this information as soon as the first Asterisk 10 beta goes out. In fact, this should occur very soon."

This transition also takes away Asterisk 2.0, and Kevin also explains why and how that came about.

"That has raised the question: Is this Asterisk 2.0? If not, will there ever be an Asterisk 2.0? After quite a lot of discussion, we came to the conclusion that this is not Asterisk 2.0, but that it’s also quite unlikely that there ever will be such a release; it wouldn’t be in the community’s best interests to release something that is fundamentally different (and not compatible) but still call it '‘Asterisk'.’ That then leaves the question we’ve been asked by many people: If there’s never going to be an Asterisk 2.0, "

Digium Blog

AstLinux 0.7.9 Is Yours For The Taking!

AstLinux Firewall

AstLinux 0.7.9 http://snapvoip.blogspot.com/
The AstLinux Team has announced the 0.7.9 release of AstLinux. This update covers several updates and bug fixes related to Asterisk. All current users are encouraged to upgrade at your earliest convenience.
AstLinux is a Linux distribution for Digium's open source IPPBX, Asterisk.

AstLinux contains powerful networking features including:

• Arno's IPTABLES Firewall
• Traffic Shaping
• IPSec, OpenVPN and PPTP VPN Support
• Caching DNS Support (dnsmasq)

Supported platforms include:

• Soekris net4801 and net5501
• VIA C3 and C7 based systems
• PCEngines ALIX and WRAP
• Generic x86 PC Hardware

The following telephony hardware cards are supported:

• Digium
• Rhino Equipment
• Sangoma
• mISDN v1

SIP Forum Releases SIPit Interoperability Test Results from SIPit 28 Held In in Huntsville, Alabama

SIPit 28 SIP Interoperability http://snapvoip.blogspot.com/

SIP Forum has released the test results from the latest SIPit event SIPit 28, the SIP interoperability event, was hosted by Digium on April 11–15, 2011 and held at the Jackson Center in Huntsville, Alabama's Research Park. SIPit is organized by the SIP Forum's Test Event Working Group (TEWG) and serves as a "plugfest" for participating companies to perform SIP interoperability testing with other participants in a live network environment. There were 19 companies from 10 countries who participated in the SIPit 28 to test and confirm SIP interoperability this spring. High definition IP video was a key focus at the event and IPv6 support in the industry was also on the rise, the forum noted.

"SIPit28 was one of the smoothest events to date featuring a lot of positive energy and hard work from all involved, We received great feedback from participating companies who were able to glean a good deal of valuable data which provided them a very high return.", said Robert Sparks, chair of the SIP Forum's Test Event Working Group and organizer of SIPit 28.said Robert Sparks, chair of the SIP Forum's Test Event Working Group and organizer of SIPit 28.

Follow the links after the jump.

Following is the press release by SIP Forum.

NORTH ANDOVER, MA (June 27, 2011) - The SIP Forum has announced the results of its latest SIP Interoperability Testing event, SIPit 28, the world's premier interoperability testing event for the Session Initiation Protocol (SIP) for application developers, service providers, and IP communications equipment manufacturers, held earlier this spring at Jackson Center in Huntsville, Alabama's Research Park. Hosted by Digium, the five-day testing event once again demonstrated the critical role that Session Initiation Protocol (SIP) plays in today's telecommunications industry while also highlighting the growth, evolution and adoption of SIP services in fixed and mobile IP networks worldwide.

SIPit 28 provided 19 attending companies with valuable "real-world" deployment data gathered from the 40 distinct SIP implementations conducted during the week as participants tested SIP interworking in a variety of simulated, live IP network conditions and settings. Testing exercised the full range of defined transport protocols for SIP, from UDP to TLS over SCTP.

As part of the week-long testing event, which took place April 11-15, participants showed a growing interest in deployment of high definition (HD) video services across global IP networks. Participants executed a number of different telepresence scenarios with intricate SIP signaling, testing for quality of service (QoS), network resiliency and interoperability involving new types of rich media endpoints with IP application servers, SIP proxy servers and session border controllers.

"Rolling out IP video in both enterprise and service provider networks and making it work is a key objective for many operators, equipment makers and application developers in today's telecom world and is something attending companies really focused on at this year's SIPit" said Robert Sparks, chair of the SIP Forum's Test Event Working Group and organizer of SIPit 28.

"We were able to successfully execute a number of testing scenarios that examined different network topologies under a variety of simulated conditions. Participants were able to verify correct behavior of advanced applications in an environment that is very difficult to duplicate without so many different implementations in the same room at the same time."

In addition, IPv6 was also a hot topic in preparation for the industry-wide rollout of this next generation Internet protocol, with many participants uncovering new, positive information surrounding interoperability of IPv6 equipment and services, particularly when implementations use a combination of IPv6 and IPv4 for media and signaling. Across the testing, 68 percent of the implementations at the event supported IPv6 which represented a significant increase from SIPit 27 in 2010 which saw 53 percent of attendees achieve IPv6 support.

A full summary report of SIPit 28 results is available at the SIP Forum website or by clicking here (https://www.sipit.net/SIPit28_summary).

"The SIPit events continue to be critically important and popular events for the IP communications industry because they give participants an opportunity to vigorously test implementations of their own IP communications product or service with other members of the telecom community in a non-commercial, real-world testing environment," said Marc Robins, SIP Forum President and Managing Director. "Every time we have a SIPit event, we receive scores of positive feedback about how valuable the testing was to the development of products and services, as well as how useful it is to standards bodies such as the IETF for gaining insight into the growth of SIP and the specific SIP implementations."

SIPit is organized by the SIP Forum's Test Event Working Group (TEWG) and serves as a "plugfest" for participating companies to perform SIP interoperability testing with other participants in a live network environment. Conducted twice a year, with events rotating in the United States, Europe, and Asia, the SIP Forum has hosted 28 events around the globe. In the coming months, the SIP Forum will release details about its next event - SIPit 29 - which will be hosted by ETSI and take place this fall in France. For information about past SIPits, please visit www.sipit.net.

"SIPit28 was one of the smoothest events to date featuring a lot of positive energy and hard work from all involved," continued Sparks. "We received great feedback from participating companies who were able to glean a good deal of valuable data which provided them a very high return."

SIPit via SIP Forum

Major Asterisk SCF Build System Changes Introduced.

Asterisk SCF Build System http://snapvoip.blogspot.com/
Yesterday Kevin Fleming, Digium's Director of Software Technologies, posted to Mailing list that he has introduced some major changes to the build system.
Asterisk SCF (Asterisk Scalable Communications Framework) was introduced to the Asterisk community at the Astricon 2010

"On Monday afternoon I merged a significant set of improvements to the Asterisk SCF build system, and along with that there were a number of changes in each of the component repositories to take advantage of the improvements. This message documents the new features (and caveats) of this improved build system:"

There is a long list of the improvements and changes, if you have not subscribed to the mailing list, follow the link below to read the post in it's entirety.
Asterisk SCF Dev

Skype (Microsoft) Drops Asterisk Support, Ending Skype For Asterisk

Skype ends Asterisk for Skype Agreement http://snapvoip.blogspot.com/
Within two weeks of acquiring Skype, Microsoft is set out to cut all Open Source ties to Skype, begining with Asterisk, the best open source IPPBX in the world.

According to a letter from Digium to it's customers, Skype will not renew its agreement with Digium that allowed Asterisk to be integrated with the Skype service through Asterisk for Skype, a product by Digium..

"It includes proprietary software from Skype that allows Asterisk to join the Skype network as a native client. Skype has decided not to renew the agreement that permits us to package this proprietary software,” according to the letter.

Skype for Asterisk sales and activations will end on July 26, but Skype has promised to continue supporting and maintaining the software for two more years. Skype may extend this time period “at their discretion.”
But I love what Digium's Danny Windham has to say about the whole issue of Skype;

Is the acquisition good for Skype? Given the valuation, it’s certainly good for the Skype shareholders. But what about the service itself? In a word, no. History suggests that Microsoft’s tendency towards notoriously proprietary tactics will slow the development of Skype as a business tool. Will Microsoft wall-off Skype from competing products, completely? Or, at a bare minimum will Microsoft ensure their products work much better with Skype than those from competitors? Time will tell.

This transaction has attracted speculation from around the globe regarding the future of Skype, Microsoft, Google, Facebook, incumbent phone companies, VoIP, video communications, Apple, the Iphone, Andriod, and I’m sure even pork belly futures. Regardless of your view on how the merger will ultimately shake out, it’s hard to deny at least this one fact: disruptive communications companies and technologies are changing the world at a rapid pace – and in this case, they are generating gigantic value for their shareholders in the process.

Digium Blog

Are You An Digium / Asterisk Innovator?

http://snapvoip.blogspot.com/
Digium just announced that they are now accepting submissions for the 2011 Asterisk Innovation Awards! The Digium Innovation Award is designed to recognize developers, customers and partners for outstanding achievements with Asterisk that are improving business processes, overcoming technology challenges and enhancing the company’s bottom line. Digium first introduced the Innovation awards at Astricon 2006 and has been an industry wide accepted award and awaited ever since.
This is open to any and all Digium|Asterisk customers and partners world-wide with solutions that are running and in production. There is also no limits to the number of projects and you can submit as many projects as long as you remember that each project requires a separate submission.

Digium will select this year’s winner based upon a number of criteria that include:

• Your description of what makes your Asterisk-based solution innovative.
• The amount to which your solution improved processes.
• The technology challenges that were overcome to achieve your goal.
• The size or breadth of your solution.
• The measurable ROI and competitive advantages generated by project.
• The things that can be achieved with the solution today that couldn’t be previously accomplished.
• The creativity demonstrated in the implementation of the project
• The overall presentation of your solution

If you happen to be the winner, you could expect to receive from Digium Asterisk team,

• Hotel accommodations (2 nights) and airfare for one
• Presentation of the award with a profile of your company in the AstriCon Conference General session
• Congratulatory press release from Digium, Inc.
• Listing on the Digium Web site
• Chance to highlight the accomplishments of you and your team
• Recognition by your industry, friends and family

When can applications be submitted? You can submit your projects to the 2011 Digium Innovation Award today! Submissions are being accepted at the following URL:
http://www.asterisk.org/innovation
Submissions are due by August 1, 2011 and winners will be announced at Astricon, 2011.

Running Your Businesses On Open Source Software!

http://snapvoip.blogspot.com/
Danny Windham's, CEO of Digium  speech at ITEXPO East 2011 in Miami about  exciting, real world, large -scale implementations of Open Source software, like Asterisk, MySQL, OpenOffice, Zimbra, Zamanda, SugerCRM and of course Apache!.

TMC

Asterisk Exchange, A Quick Q&A With Digium Community Director Bryan Johns

Asterisk Exchange http://snapvoip.blogspot.com/
The Var Guy has conducted what he calls a Quick Q&A, but I think very focused Q&A with Digium Community Director Bryan Johns to extract some facts about the Asterisk Exchange alliance.
Asterisk Exchange has attracted more than 50 partners ranging from hardware providers, ISVs (independent software vendors) and VARs.
At the end of the Q&A session, the Var Guy offers his own views on the Asterisk Exchange. A good read.
The Var Guy

Digium 2010, What A Year It Has Been!

Digium 2010 http://snapvoip.blogspot.com/

Digium Blog

Asterisk SCF, A New Open Source Telephony Project, Will Simplify Creation of Complex Communications Systems

Asterisk Scalable Communications Framework. Asterisk SCF http://snapvoip.blogspot.com/
Digium, the asterisk company announced the Asterisk SCF, Asterisk Scalable Communications Framework at the Astricon 2010. Digium along with the Asterisk Open Source community will work together to provide real time communication solutions encompassing voice, video and text to accommodate today's needs of  always online community, business or otherwise.
The Asterisk SCF platform will support, as a part of its basic architecture, the full range of real-time IP communications, including video, multi-channel wideband and ultra-wideband audio, chat, desktop sharing and other media types that may arise in the future.
"The Asterisk community asked for easier ways to use Asterisk in larger and more complex applications, They want to make massive scalability and fault tolerance simple, they want rich APIs for developing applications, and they want performance that effectively utilizes modern systems and architectures. Rather than trying to 'grow' Asterisk to address these areas, Digium and a committee of Asterisk developers decided that a companion product would be the most effective approach, since it allowed us to consider the widest range of tools, platforms and architectures." said Mark Spencer, creator of Asterisk and Digium's founder and chief technology officer.
A version of Asterisk SCF for early adopters is now available at the Asterisk website (http://www.asterisk.org). The project will evolve aggressively during the next 12 months. Digium invites developers interested in contributing or wanting to learn more about Asterisk SCF to visit http://www.asterisk.org today.
The Asterisk community and the Digium also pointed out that this does not mean the Asterisk SCF is not a replacement for the Asterisk, the world's most widely used open source voice communications platform. Digium and the Asterisk community are committed to the continued development and support of Asterisk, the telecommunications software.  This commitment is emphasized by the recent announcement of the release of the next long-term support version of Asterisk, Asterisk 1.8 LTS.

Asterisk 1.8.0 RC3 Released For Testing And Reporting.

#Asterisk 1.8.0 RC3 Released For Testing And Reporting. http://snapvoip.blogspot.com/
The release candidate of Asterisk 1.8.0. has been released by the Asterisk Development Team this release candidate 3, RC3 is available for immediate download from Asterisk.Org
Since any beta or pre release issues are tracked by greater participation of the Asterisk Community and all are invited and encouraged to participate in the 1.8 testing process. Please report any issues found to the Asterisk issue tracker. It is also very useful to see successful test reports those could be posted to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk and is slated to be a Long Term Support (LTS) release, just like the very successful,  Asterisk 1.4. You can and should learn about Asterisk versions at Asterisk versions page.

The binary add-on modules for Asterisk produced by Digium have been updated to be compatible with Asterisk 1.8. The availability of these modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations and usage scenarios.

The release candidate RC3 contains fixes since the release candidate as reported by the community. A sampling of the changes in this release candidate include:

    * Still build chan_sip even if res_crypto cannot be built (use, but not depend) (Reported by a user on the mailing list. Patched by tilghman)    * Get notifications for call files only when a file is closed, not when created (Closes issue #17924. Reported by mkeuter. Patched by abeldeck)
    * Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk expects the DTMF to arrive on the RTP stream and not via jingle DTMF signalling.(Patched by dvossel. Tested by malcolmd)
    * Fixes to allow chan_gtalk to communicate with the Gmail web client. (Patched by phsultan and dvossel)
    * Fix to GET DATA to allow audio to be streamed via an AGI. (Closes issue #18001. Reported by jamicque. Patched by tilghman)
    * Resolve dnsmgr memory corruption in chan_iax2.(Closes issue #17902. Reported by afried. Patched by russell, dvossel)

Some of the notable features includes:

    * Secure RTP
    * IPv6 Support in the SIP channel driver
    * Connected Party Identification Support
    * Calendaring Integration
    * A new call logging system, Channel Event Logging (CEL)
    * Distributed Device State using Jabber/XMPP PubSub
    * Call Completion Supplementary Services support
    * Advice of Charge support
    * Much, much more!

A full list of new features can be found in the CHANGES file.
For a full list of changes in the current release candidate, please see the ChangeLog:

"Future of Open Source Communications", A Keynote By Digium At AstriCon 2010.

"Future of Open Source Communications" @ #Digium  #Astricon http://snapvoip.blogspot.com/
Digium revealed today that Kevin Fleming, director of software technologies, will be delivering a keynote speech cum presentation regarding the future of open source communications software at the AstriCon 2010, which will be the  seventh annual AstriCon Open Source Telephony Conference and Exhibition, taking place from October 26-28, 2010, at the Gaylord National Resort and Convention Center in National Harbor, Maryland.
I plan to be at the conference and I will attend presentation. Fleming's keynote takes place on Wednesday, October 27 at 9:00 a.m. ET. This is not the first such keynote and seem to be a tradition of AstriCon. The last was presented by Sean Michael Kerner, "The Future of Open Source VoIP is Asterisk 1.6" in 2007.

As we all know, Asterisk is the forefront of Open source telephony and I have been attending the conference every year since the beginning. This is the place where telephony developers, telephony and unified communications experts, resellers and VARs, and Asterisk enthusiasts come together to gain and share knowledge.

"Asterisk has been an industry-changing open source project that has grown with the support of an active and thriving community, We believe that the news we're sharing at AstriCon will further the disruption of the communications industry, as Asterisk did with its original release, and demonstrate Digium's commitment to taking open source communications in an exciting direction for the Asterisk Community." said Fleming, who has helped manage and define Asterisk's direction since he joined Digium in 2005

You can find additional information and open registration are available at http://www.astricon.net.
Digium

Announcing AstriDevCon 2010 At AstriCon 2010!

AstriDevCon Reestablished http://snapvoip.blogspot.com/John Todd has posted to Asterisk-Dev mailing list about re-establishing AstriDevCon, the mini-conference within AstriCon 2010 conference. I am sure many developers will find it most welcoming as they get to spend one whole day with fellow Asterisk core developers;

where core developers can have direct and uninterrupted access to each other in a more direct way than usual.These sessions often result in massively productive results, either as new features are contemplated and developed, or as old problems are finally vanquished through the application of many eyes and hands. Says John Todd

Seats are limited.  Preference will be given to developers who are active participants in the issue tracker, patch submission, and asterisk-dev mailing lists and IRC channels.  There is no charge for attending AstriDevCon, though registration is required. Register for AstriDevCon at http://bit.ly/astridevcon2010
John also mentions that the AstriDevCon is fairly unstructured, but it is typically the case that there are discussions on:

   - security issues
   - priority of work issues
   - general brainstorming on new features
   - difficult bugs in the issuetracker
   - protocol (SIP, IAX2, etc.) questions and decisions
   - integration of third-party applications

If you have any questions, get in touch with John Todd via the email address provided at the following link
Asterisk-Dev@Digium

Digium® Certifies Panasonic line of SIP Cordless Phone Systems, the KX-TGP500 and KX-TGP550, for Asterisk open source PBX

KX-TGP500 and KX-TGP550, for Asterisk open source PBX http://snapvoip.blogspot.com/
SECAUCUS, NJ--(Marketwire - May 4, 2010) -  Panasonic System Networks Company of America, the leader in business and home telephone systems, today announced Digium®, the Asterisk® Company, has certified the Panasonic line of SIP Cordless Phone Systems, the KX-TGP500 and KX-TGP550, for use with the Asterisk telephony platform. By leveraging Panasonic's market leadership in DECT cordless telephones and Digium's innovative open source PBX, this alliance is a winning combination.
The Panasonic TGP500 series of SIP Cordless Phone Systems boasts outstanding voice quality and a range of productivity-boosting options in a feature-rich desk phone replacement. DECT 6.0 ensures no interference with wireless networks, and the convenient cordless design eliminates the need to run dedicated network wiring to each employee workstation. It is ideal for home and business environments.
Digium is the creator, sponsor and driving force behind Asterisk, the most widely used open source telephony software. The company's product lines include a wide range of software and hardware that enable businesses to implement turnkey unified communications solutions or to design their own voice-over-IP (VoIP) systems. Resellers, telecom professionals and software developers choose Digium's products because only Digium delivers the technical superiority, security and flexibility associated with Asterisk.
"We are excited to work with Digium and their fast growing base of customers using open source, standards-based business phone systems," said Kim Murtagh, Director, Panasonic System Networks Company of America. "As the market leader, Panasonic is committed to delivering innovative features and superior performance."
"The Digium and Panasonic interoperability partnership and certification of IP phones opens new opportunities for Digium channel partners and integrators," said Digium's David Hardy, Director of Business Development. "The Panasonic global footprint and commitment to SIP telephony excellence creates exciting sales opportunities for our partner community and for businesses using Asterisk."
With flexible configuration options, it has never been easier to deploy and expand a SIP-based phone system. Handsets can be easily added wherever needed; all that is required is an electrical outlet nearby. The benefits of SIP are especially compelling in today's business environment, where every dollar counts. The reduced hardware costs and simplicity of routing calls over an Internet connection can add up to huge savings on monthly telephone bills. With the Panasonic TGP500 series SIP phone system, it is quick and easy to add up to 6 cordless handsets -- each with its own number. All that is needed is a single Internet connection and an electrical outlet near the location of the handset. Because it employs DECT technology, the connection is secure and the sound crystal-clear.
TGP500 Series Details

KX-TGP500
The system features a wall-mountable base unit and one cordless handset. It is expandable up to 6 DECT 6.0 cordless handsets and supports up to 8 phone numbers and 3 simultaneous calls. It boasts Wide Band Audio (G.722) and 5 hours Talk Time, 10 days Standby. Its elegant design features a white backlit large LCD on the handset and a Handset locator button on the base unit. It also has a handset speakerphone, 2.5mm headset jack and belt clip.

KX-TGP550
The KX-TGP550 has all the features and benefits of the KX-TGP500 and adds a corded base unit with a large white backlit LCD and 5 hours Talk Time, 10 days Standby, plus a Hands-Free Speaker phone, Handset Call Button on the base unit, and one-touch call transfer with Busy Lamp Indication.

KX-TPA50
The TGP500 systems can be expanded up to a total of 6 cordless handsets by adding the KX-TPA50 cordless handset.
The TGP500 Series will be available in May 2010 through the following Panasonic authorized distribution partners: Jenne Distributors, NETXUSA and D&H Distributing.

A Case For Switchvox, Another Case For Asterisk, Which One Is Yours?

The Case For Asterisk and Switchvox snapvoip.blogspot.com
Digium blog has published a very clear and nice article by Steve Sokol that speaks about virtues of both the products. Asterisk is the industry favorite Opens source telephony project and the SwitchVox is a Unified Communications system which is also based on Asterisk. The former is free and the latter is not. But there are many reasons that you may want to choose one or the other. The aricle basically ends with a clear advise;

If you’re technically inclined and want to build a communication product or solution, then Asterisk is for you.  If you’re in need of a great phone system at a great price, check out Switchvox.

Asterisk or Switchvox?

Best Practices as applied to implementation of an Open Source IP Telephony Solution.

Open Source IP Telephony snapvoip.blogspot.com
Bill Miller (beelinebill) of Digium has published the third and final part in a series of Open Source policy for IP Telephony. I have enjoyed his previous posts and this one is no different.
You can access all three articles from the link below, which covered;

1. Introduction to Open Source Policy for IP Telephony in the Enterprise:  “Open Source Policy for IP Telephony:  Are You Behind or Ahead of the Curve?”
2. Business Analysis of Open Source vs. Traditional Telephony: “The Metrics of Open Source Policy for IP Telephony “
3. Best Practices as applied to implementation of an Open Source IP Telephony solution: “Open Source Policy Meets ROI”

Beelinebill

Skype CEO Keynotes ITEXPO East 2010, Video

ITEXPO East 2010 snapvoip.blogspot.com
Rich Tehrani has posted a video of Skype CEO,  Danny Windham Keynote at ITEXPO East 2010, on his blog. If you missed the ITEXPO East 2010, then it is something you should watch.
Windham seem to have the 10 Year rise of Digium compared to the fall of Nortel. Well he says Open Source is to blame or applause, depending on which camp you are in!
I am of course applausing!
Rich's blog

Fax For Asterisk 1.1.6 Released!

Fax For Asterisk 1.1.6 snapvoip.blogspot.com
Digium has announced the release of Fax For Asterisk 1.1.6 for open source Asterisk. Digium attests that this a a commercial grade FAX add-on module.

The latest release brings number of improvements;

* Support for 64-bit Linux installations.

* Reduction in resource consumption, and increase in performance, of T.38 session handling.

* Simplification of session handling during transition from G.711 to T.38 mode.

* Adoption of the latest Asterisk T.38 negotiation API, ensuring interoperability with a wide range of T.38 endpoints.

The Fax For Asterisk 1.1.6 is could be downloaded from Digium servers (http://www.digium.com/en/docs/FAX/faa-download.php).
Note: Because this release uses the very latest T.38 negotiation mechanism in Asterisk, you need to check the Fax For Asterisk download selector lists for the Asterisk versions supported by the latest release.
Fax For Asterisk  product page.

Asterisk Branding!

Asterisk Branding, snapvoip.blogspot.com
I had the pleasure of reading an article by TMC's Erik Linask interviewing Digium's in-house counsel  and contracts manager Michelle Petrone-Fleming (whom we encountered in her Asterisk, Open Source and Licensing! article)concerning Digium and Asterisk branding and regarding the presentation that Michelle will be presenting at Astricon 2009.
Great read, and Michelle mentioned at one point;

“If you show a random sampling of code to anyone not intimately familiar with every line of code in Asterisk, and they will have a hard time telling you what software it is. Show that same person the bubble logo and they’ll immediately know it is Asterisk.”

Yes.
TMCNet