Magento CE Version 1.6.0.0 Stable – Now Available
We are excited to announce the availability of Magento CE Version 1.6.0.0 Stable for download and upgrade.
The latest release is packed with new features as well as valuable code contributions from various community members around the world.
Some of the key new features in this release include:
- Persistent shopping - retain shopping cart content for customers across user sessions, browsers and devices.
- Minimum Advertised Price (MAP)
- Refactoring multiple database
To see the full list of features and fixed issues please visit our release notes page. Diff files are available here.
Please Note: We do NOT recommend upgrading a production installation of Magento directly. Please backup database and all files before upgrading. Please make sure to check file permission before trying to upgrade through your Magento Connect Manager.
Please report all issues with this release in the bug tracker.
Source: http://www.magentocommerce.com/blog/comments/magento-ce-version-1600-stable-now-available/
Magento CE Version 1.5.1.0 Stable – Now Available
We are excited to announce the availability of Magento CE Version 1.5.1.0 Stable for download and upgrade.
To see a full list of features and fixed issues please visit our release notes page. Diff files are available here.
Please Note: We do NOT recommend upgrading a production installation of Magento directly. Please backup database and all files before upgrading. Please make sure to check file permission before trying to upgrade through your Magento Connect Manager.
Please report all issues with this release in the bug tracker.
Source: http://www.magentocommerce.com/blog/comments/magento-ce-version-1510-stable-now-available/
Magento Developers Paradise: Ibiza, Spain
Magento Developers Paradise
June 4-7, 2011
The early bird registration rate ends on Sunday April 10th!
An awesome 3-day program dedicated to the Magento developer community is starting to shape up. This year we are meeting in Ibiza, Spain. Your opportunity to leave the office, exchange ideas, learn from other Magento developers and have some fun is just around the corner.
You’ll Never Forget It!
The event promises to provide some unique and memorable experiences. Certainly an event you don’t want to miss. We promise:
- Lots of Code
- Lots of Ideas
- Fun in the Sun
- Mingle with the Best and Brightest
This is a great 3-day event to learn and exchange ideas about Magento development, work with the Magento code base, build extensions and get a close and intimate look at what our developers are working on.
Be sure to sign up this week to take advantage of the early bird rates with savings of €200.
For complete and up-to-date conference information please visit the Meet Magento Event websitehttp://www.magento-developers-paradise.com/home.html
Source: http://www.magentocommerce.com/blog/comments/magento-developers-paradise-ibiza-spain/
Home Run: Asterisk Baseball Scores & Schedules with Gtalk
Last week we introduced the new Worldwide Weather Station for Asterisk 1.8 using Google's new Google Talk Guru. And, as promised, today we bring you the first of several new Asterisk applications to retrieve sports scores and schedules from the convenience of your telephone. With Google Talk Guru, you can retrieve the latest Atlanta Braves info by issuing this Chat command:score braves. What you'll receive in reply using Google Chat within Gmail would look something like this:
Baseball:
Atlanta Braves 2 - Milwaukee Brewers 1
Next game: @ Milwaukee Brewers, 6 Apr 3:10am
mlb.mlb.com
With today's installation, you'll also be able to dial M-L-B (652) from any Asterisk extension and retrieve the latest score and next game schedule for any one of 10 Major League Baseball teams by pressing a single button. For example, to retrieve the latest Atlanta Braves score and next game schedule, press 2. To try out our demo, just dial 425-406-4532 from any phone in the U.S. Here's the entire list which you can modify to meet your own requirements:
0 - Yankees
1 - Mets
2 - Braves
3 - Reds
4 - Marlins
5 - Orioles
6 - Pirates
7 - Royals
8 - Dodgers
9 - White Sox
As was true with weather forecasts, retrieval of baseball scores and schedules using Google Talk Guru takes less than a second for almost any team. And, in addition to playing these scores and schedules over the phone using Asterisk 1.8, we've added the ability to also forward the results to your favorite email address. If you're already familiar with last week's installation procedure, then drop down to the Quick Installation topic. The whole drill should take you no more than a couple minutes. If you're new to all of this, keep reading.
How It Works. Here's a quick summary of how all this works. With the Google Talk Guru, you can send a query as a text message to guru@googlelabs.com. You then get a reply message in Google Talk with the answer to your query. What we've done is add this querying functionality to your Asterisk dialplan with some preassigned baseball teams to obtain the latest sports scores and schedules. Once the response arrives, we've added a PHP application that puts the text (as shown above) into something that's a little more TTS friendly for Flite and Cepstral. If you're curious about how to do all of this, take a look at the dialplan and PHP code in the links below. It's not hard, but it is tedious. One little typo and nothing works. Ask us how we know. 
Prerequisites. If you're new to all of this, here's a quick list of what you'll need. First, you'll need a PBX in a Flash server running the very latest Asterisk 1.8. We call it PIAF-Purple. Bidirectional Google chatting only works in the most recent releases of Asterisk 1.8 so, no, you can't wing it with an earlier release and expect a working system. Next you'll need to add Google Voice and Chat support. You can install these components yourself, or you can use Incredible PBX 1.8. The latest release as of today has this application preinstalled. If you dial 652 from an extension on your Incredible PBX and are prompted to choose a team for the latest score and schedule after hearing a list of the available teams, then your installation is complete even though it won't work until you invite yourself to chat with guru@googlelabs.com using the same Gmail account you're using for Google Voice on your Asterisk server. If dialing 652 doesn't work, then you'll need to add this application to your existing Incredible PBX 1.8installation by following the simple steps below in addition to enabling chats with guru@googlelabs.com. Almost any other (current) Asterisk 1.8 server should work as well so long as you've installed FreePBX, PHP and the Flite or Cepstral voice synthesizer. But then you're on your own. If you're a nuts-and-bolts Asterisk guy, then you should be able to decipher what needs to be done by reading through this tutorial.
Quick Installation. Assuming you have all the prerequisites in place, today's installation is about a five minute chore. There are 3 easy steps:
(1) While signed in to Gmail with the same account credentials being used for Google Voice on your Asterisk server, activate chat temporarily and invite yourself to chat with guru@googlelabs.com. Run a test query using the Braves example above. IMPORTANT: Once it works, disable chat on your desktop, or Google Voice and Chat will no longer work with Asterisk!
(2) Download the Baseball Scores & Schedules application into the agi-bindirectory on your Asterisk system. Here are the commands after logging into your server as root:
(3) While still logged in as root, switch to the /etc/asterisk directory and edit extensions_custom.conf with this command:
nano -w extensions_custom.conf
Search for 652 and delete any existing lines with that extension. Then cut-and-paste the following code inserting it just below the [from-internal-custom] context marker (but above any other context marker) or in the existing position if you deleted existing 652 lines. Use nano -w extensions_custom.conf to open the file, or word wrap will delete part of the cut-and-paste code! Once you've saved your changes, reload your Asterisk dialplan:
asterisk -rx "dialplan reload"
Customization. By default, the application is set to use Flite as the text-to-speech (TTS) engine. If you have installed Cepstral, you can change to Cepstral. In the /var/lib/asterisk/agi-bin directory, edit nv-mlb-google.php and change$ttspick = 0 to $ttspick = 1. Do not delete the trailing semicolon! If you want the sports scores and schedules also emailed to you when you dial them up, then insert your actual email address in the $email variable and set$emailscore = 1.
You need not use the 10 teams that are preconfigured in the application. You can choose your own. First, write down the names of the 10 teams you wish to use. Do NOT use city names! Make a backup of extensions_custom.conf: cp extensions_custom.conf ext_custom.bak. Then carefully edit /etc/asterisk/extensions_custom.conf using nano -w filename. Move down to the 652,3 and 652,5 lines and make the necessary changes using the teams you have chosen. Finally, move down to 652,50 and replace Yankees with your 0 choice, 612,52 Mets with your 1 choice, etc. Save your changes and reload your dialplan. NOTE: For multi-word teams such as White Sox, be sure to use an underscore between the words, NOT A SPACE, e.g. white_sox.
If you want to retrieve scores and schedules for more than 10 teams, the easiest solution is to clone all of the 652 dialplan code and renumber each occurrence of 652 to 653. HINT: Some 652 entries are actually embedded in the code as well as in the extension numbers. Be sure to renumber those entries as well. Use Ctrl-W to find each 652 occurrence in the new context, and you won't inadvertently miss one. That gets you 10 more teams. Repeat as desired. Note also that you need not announce 10 teams in the voice prompt unless you want to. If you only plan to follow 3 teams, then alter the initial voice prompt to only announce those teams. You do NOT need to delete the dialplan code that actually picks other teams. No one will ever know.
Adding a Miscellaneous Destination. This step is optional. Access FreePBX with your browser, and choose Setup, Misc Destination. If it's not already there, add a new entry for MLBScores with 652 as the Dial entry. Save your entry and then click the Red Bar to reload Asterisk.
Taking Baseball Scores and Schedules for a Spin. Now we should be all set. Just pick up an extension on your system and dial 652. You'll be prompted to enter a one-digit code. Punch in 5 and check out the latest score and next game for the Baltimore Orioles. Enjoy!
Housekeeping 101. Temporary files in /tmp get cleaned up by Linux housekeeping automatically. Temporary files stored elsewhere don't unless you're using Incredible PBX. The weather scripts store .wav files with your requested weather forecasts in /var/lib/asterisk/sounds/tts. So, from time to time, make a mental note to remove all of these files with a command like this:
rm -f /var/lib/asterisk/sounds/tts/tts*
Or just log into your Asterisk server as root and edit the following file: nano -w /etc/crontab. Move to the bottom of the file and insert the following code on a blank line:
01 0 * * * root rm -f /var/lib/asterisk/sounds/tts/tts* > /dev/null
This code will delete all of the TTS files in the tts folder every night. Now save your changes: Ctrl-X, Y, then Enter.
Best of Nerd Vittles Link. This application also will be available on our Best of Nerd Vittles site shortly. Enjoy!
Source: http://nerdvittles.com/?p=735
Magento CE Version 1.4.2.0 Stable – Now Available
We are excited to announce the availability of Magento CE Version 1.4.2.0 Stable for download and upgrade.
To see a full list of features and fixed issues please visit our release notes page. Diff files are available here.
Please Note: We do NOT recommend upgrading a production installation of Magento directly. Please backup database and all files before upgrading. Please make sure to check file permission before trying to upgrade through your Magento Connect Manager.
Please report all issues with this release in the bug tracker.
Source: http://www.magentocommerce.com/blog/comments/magento-ce-version-1420-stable-now-available/
Open Source VoIP the Right Way
By TMCnet Special Guest
Frederic Dickey, Director of Marketing and Product Management at Sangoma
The implementation of open source voice over IP solutions has graduated from niche status to become a viable and increasingly popular approach for enterprise voice services. A 2009 research study by the Eastern Management (News - Alert) Group reported that open source PBXs captured 18 percent of the 2008 market – a 40 percent increase over the previous year, and more market share than any traditional PBX manufacturer.
Platforms based on Asterisk (News - Alert) and FreeSwitch offer IT managers comparable features to traditional telephony systems and prove more cost effective and easier to deploy and maintain. Despite the simpler and less costly approach, however, certain steps must be to taken to ensure maximum scalability and optimum voice quality in deployment of an open source VoIP system. Because VoIP runs on data networks, because open source telephony systems run on standard computing systems, and because these systems run on standard OSs, more careful attention must be paid to addressing capacity and quality issues that are either not as prevalent or not considerations at all in traditional TDM implementations.
Putting voice on a packet network and running telephony systems on standard computers change the rules of the game. As a result, there are several important hardware and software considerations to make to make sure all quality and capacity issues are addressed and the benefits of open source VoIP are fully realized.
Determining Optimal Payload SizeVoice traffic is placed into a packet network using the real-time transport protocol, or RTP. The nature of packet networks is such that a certain length of speech must be accumulated before sending out the packet, which consumes the processing power of the CPU. This means that determining optimal payload size can be a tricky task. If payload size is too large, the quality of the conversation can be adversely impacted because of the introduction of too much delay –which can result in awkward conversations with people talking over one another.
Typically, a range of between 25 and 150 milliseconds end-to-end delay is acceptable. To account for other delays such as jitter and voice compression/decompression, 30 milliseconds tends to be the upper limit. That means that the computing platform will create RTP packets for every single conversation at 30-millisecond intervals, ensuring a smooth conversation.
The Importance of Echo Cancellation
Echo is prevalent in any telephone network. In traditional telephony networks, echo was typically an issue only for overseas calls, or calls with roundtrip delays of more than 30 milliseconds. But in the case of VoIP, delays for calls are typically longer, which makes echo more perceptible to the human ear – so good echo cancellation is required on all calls, even if the call is between colleagues who are in cubicles next to one another.
Echo cancellation algorithms are very complex and processing intensive. While they can be implemented in software, this task is more appropriately addressed by utilizing specialized DSPs on telephony boards that can handle the load, which frees up the computing platform to process other important tasks. The VoIP phones selected for an open source implementation should be equipped to address echo cancellation as well, including the reduction of acoustic echo.
Selecting the Right Hardware
To prepare for an open source VoIP deployment, the configuration of the servers used to run the system must first be addressed. To ensure a scalable, reliable deployment, CPU occupancy and memory should be run in steady mode under 60 percent load to allow for peaks in traffic. RAID storage solutions should be utilized to maintain continuity and ensure adequate capacity to allow for functions like call data records and voicemail recordings.
Most standard computing systems do not come equipped with telephony interfaces, so accommodating an open source VoIP deployment means purchasing and installing PCI (News - Alert) or PCIe boards. These boards are not created equal, so careful consideration should be given to ensure an optimal deployment with the right features and functionality.
Telephony boards must continuously transfer voice buffers to the host CPU, so having the ability to configure the frequency at which this is done will help optimize the performance of the CPU. Some boards are hardwired to interrupt the CPU at every millisecond for every call to perform this task, while others are configurable for longer periods so that the CPU can be relieved to address other tasks. As with RTP packet sizes, there is a fine balance to keep between CPU occupancy and delays introduced.
On-board echo cancellation and on-board tone detection are other functions that take processing power away from the CPU and must be configured correctly. In addition, on-board HDLC framing, which is required to pass call control protocol data such as ISDN D-channels from the network to the host, consumes CPU cycles.
There are many other considerations, such as network design, on which we have not touched. But clearly, deploying an open source VoIP systems introduces a lot of changes and requires careful consideration of many network settings and functions, as well as the right selection of hardware and computing platforms. When deployed correctly following these guidelines, VoIP systems based on open source can provide high-quality and efficient voice communication for enterprises.
TMCnet publishes expert commentary on various telecommunications, IT, call center, CRM and other technology-related topics. Are you an expert in one of these fields, and interested in having your perspective published on a site that gets several million unique visitors each month? Get in touch.
Edited by Stefania Viscusi
Source: http://ip-telephony.tmcnet.com/topics/ip-telephony/articles/115610-open-source-voip-right-way.htm
The Incredible PBX: Safely Interconnecting Asterisk Servers
WOW! What a couple of weeks it has been. The response to Incredible PBX for Asterisk 1.8 has been, well, incredible. Just last week, SlickDeals and FatWalletintroduced over 50,000 bargain hunters to the beauties of Asterisk and Google Voice using Incredible PBX. They joined our regular 50,000 weekly visitors in discovering what may be the best VoIP calling platform on the planet, free or otherwise.
But we’ve also heard from long-time users of PBX in a Flash: “How can we take advantage of this new Google Voice technology without breaking our existing server?” Well, starting today, it’s easy! We’re going to show you how to interconnect as many Asterisk servers as you like using a simple FreePBX tweak to make free calls using your Incredible PBX. To begin, just set up a second server or virtual machine running Incredible PBX 1.8. Then we’ll walk you through interconnecting it with any other Asterisk server that’s running FreePBX. It really is a 5 minute project… once you’ve finished reading this article.
Don’t be intimidated by all of the screen shots shown below. We’re just showing multiple ways of doing the same thing. So you don’t need to use all of them. Once you’ve added one trunk entry on each of your servers and an outbound route on your existing Asterisk server, all of the users on your primary server can instantly begin making free outbound calls through the Google Voice setup on your Incredible PBX. Keep in mind that, at least for now, there is no limit to the number of simultaneous (free) outbound calls you can make within the U.S. and Canada using the Incredible PBX 1.8 platform. And you can interconnect as many Asterisk servers as you like assuming you have the 100kbps VoIP bandwidth to support each simultaneous call.
To get started, follow our last article to get an Incredible PBX 1.8 server set up. As shown in the diagram above, we’re going to assume you’ve got both your new and old Asterisk servers running on the same subnet behind a very secure hardware-based firewall. But this isn’t really required from a technical standpoint. One or more additional servers could be strung all around the globe if that’s your requirement. Or you may wish to take advantage of the incredible deal at RentPBX.com and let them host Incredible PBX 1.8 for you at $15 a month. Just use this special coupon code: BACK10. Then all of your other Asterisk servers can take advantage of today’s free-calling solution. We would hasten to add that, once you’re using the Internet as the transport mechanism for interconnecting servers, we recommend you read and use the secure VPN setup outlined in our VPN in a Flash knol, but the IAX setup outlined below is secure except your voice data is not encrypted. So that’s your call to make.
Today’s Drill. We’re going to show you how to make calls from your existing Asterisk server through The Incredible PBX today. We’ll leave it to you to get things working in the other direction if that is a requirement for your project. First, we’ll create a new trunk on The Incredible PBX, and then we’ll create both a new trunk and a new outbound route on your existing server. We’ll also cover two different interconnection setups. First, we’ll do it using SIP. And then we’ll show you a similar setup using Asterisk’s IAX.
If both servers are sitting on the same private LAN, then the SIP setup is a little easier because the Linux firewall running on Incredible PBX allows SIP traffic to flow freely without any adjustment. It assumes you have added the recommended hardware firewall layer of protection with SIP access to your servers closed off. If one or more of your servers are outside the hardware firewall that is protecting Incredible PBX 1.8, then we recommend the VPN solution referenced above first and the IAX solution outlined here as a second option because the data is unencrypted. Both of these options avoid having to open up any SIP ports on your hardware firewall, and require only a minor adjustment to IPtables, the Linux-based firewall running on The Incredible PBX.
Naming Conventions. To keep things simple, we’re going to refer to the two servers in our example as incredible-pbx and piaf-main where incredible-pbxis your new Incredible PBX 1.8 server that will host the outbound Google Voice calls for users on your piaf-main server. You can obviously adjust these names in any way you like. The only gotcha is that Asterisk attempts to match an incoming call’s username against one of its corresponding trunk names before allowing the call. If there’s no match, the call will fail. So make sure that, if you change the names in the example, do it for both the username and trunk name entries on both servers. Better yet, follow the naming convention in our example, and it just works.
Setting Up Incredible PBX for Interconnecting Servers. Let’s set up a SIP and IAX trunk on your Incredible PBX first. You really don’t need both of these. To repeat, if The Incredible PBX is located on the same private subnet as your other Asterisk server, just use the SIP trunk. If you need access from an Asterisk server outside your private LAN, use the IAX setup. To begin, login to FreePBX using maint and the password you set up with passwd-master. To create a trunk, first choose Setup, Trunks.
To create a SIP trunk, click Add SIP Trunk. For the Trunk Name, enter piaf-main. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration String blank, and clickSubmit Changes. Replace 192.168.0.50 with the actual IP address of your piaf-main server. Replace password with a very secure alphanumeric password. Leave the other entries as they are.
To create an IAX trunk, click Add IAX2 Trunk. For the Trunk Name, enter piaf-main. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration String blank, and click Submit Changes. Replace 192.168.0.50 with the actual IP address of your piaf-main server. Replace password with a very secure alphanumeric password. Leave the other entries as they are.
With either or both trunks, you have the option of tightening up how calls placed from the other server are routed. To force all calls to go out through the Google Voice trunk, just change context=from-internal to context=gvoice. If you want extensions on the other server to be able to call extensions on The Incredible PBX directly, leave the context entry the way it is shown.
While we don’t recommend it, if you’re going to have multiple Asterisk servers connecting to The Incredible PBX to place Google Voice calls and you’re too lazy to create separate trunks to support each server, you can eliminate the IP address checking mechanism in Asterisk by replacing host=192.168.0.50 withinsecure=port,invite. The security implications should be obvious.
Setting Up The Other Asterisk Server. There are two steps in setting up any other server that you wish to interconnect with The Incredible PBX. First, you have to create a compatible trunk to handle the calls. Then we’ll add an Outbound Route to send certain calls to Incredible PBX for processing. If you’re using SIP on the Incredible PBX, then you have to use SIP on the other Asterisk server. Same goes for IAX. We’ll set up both a SIP and IAX trunk on the PIAF main server just to show you what the entries should look like. And, to repeat, you really don’t need both of these. If your other Asterisk server is located on the same private subnet as Incredible PBX, use the SIP trunk. If you need access to Incredible PBX from elsewhere, use the IAX setup. To begin, login to FreePBX on your other PIAF server using maint and the password you set up withpasswd-master. To create a trunk, first choose Setup, Trunks.
To create a SIP trunk, click Add SIP Trunk. For the Trunk Name, enterincredible-pbx. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration Stringblank, and click Submit Changes. Replace 192.168.0.212 with the actual IP address of your incredible-pbx server. Replace password with the same secure alphanumeric password you used on the Incredible PBX SIP trunk to which you will be connecting. Leave the other entries as they are.
To create an IAX trunk, click Add IAX2 Trunk. For the Trunk Name, enterincredible-pbx. Then skip down to the Outgoing Settings and use the following as a guide. Then clear out the Incoming Settings, leave the Registration Stringblank, and click Submit Changes. Replace 192.168.0.212 with the actual IP address of your incredible-pbx server. Replace password with the same secure alphanumeric password you used on the Incredible PBX IAX trunk to which you will be connecting. Leave the other entries as they are.
You’ll notice in the Dial Rules, we’ve used 48 (which is GV on a phone) as the prefix to be dialed on your other Asterisk server to route calls out through Google Voice on The Incredible PBX. So, to place a call from your other Asterisk server via Google Voice, a user would dial something like this: 48-678-987-6543. Before the call leaves the Asterisk server, the 48 prefix will be stripped off. You can make this prefix anything you’d like. Just be sure to use the same prefix when you set up the Outbound Route in the next step.
Adding an Outbound Route. The final configuration step is to add a new outbound route on your other Asterisk server to actually send calls to The Incredible PBX. As noted, we use a dialing prefix so that we can identify the calls to be sent. Create a new route called GoogleVoice and make your entries look like the following if you’re using IAX. If you’re using SIP, just change Trunk Sequence 0 to SIP/incredible-pbx. Click Submit Change and reload FreePBX when prompted.
Keep in mind that FreePBX processes Outbound Routes in top down order, and the first matching route is the only route that is used to place the call even if the call fails. So the trick here is to move your new GoogleVoice route up the list so that it’s at least above the default calling route (which is a route with no specified dial patterns to match) and any other routes consisting of 12 or 13-digit dial strings which might match our GoogleVoice dial patterns.
IAX Firewall Adjustments. If you’re using the IAX method above, you’ll need to adjust the IPtables firewall rules on Incredible PBX to allow communications with your other Asterisk server. If your other Asterisk server is PBX in a Flash, you may need to add a similar entry in the IPtables rules on that machine as well. In addition, you’ll need to map UDP 4569 on your hardware-based firewall to the private IP address of your Asterisk server. Otherwise, calls will never make it past your firewall.
On each server, edit /etc/sysconfig/iptables and add an entry with the IP address of the other server with which you’ll be communicating. If your Incredible PBX is on a different public network than your other server, we’d need to add an entry near the end of the file and above COMMIT allowing IAX communications with the public (not private!) IP address of the piaf-main server assuming that server is outside the LAN, e.g. something like this:
-A INPUT -p udp -m udp -s 222.68.100.150 –dport 4569 -j ACCEPT
If you’re using IAX and both servers are on the same private subnet or interconnected private subnets, then the entry might look like this:
-A INPUT -p udp -m udp -s 192.168.0.50 –dport 4569 -j ACCEPT
Once you’ve saved your change, restart the firewall: service iptables restart
Testing Things Out. Now you’re ready to place a test call. Pick up an extension on your piaf-main system and dial 48-800-322-7300. You’ll be greeted by American Airlines courtesy of Google Voice. The CallerID of your outbound calls will be your Google Voice number regardless of the extension or server from which the call originates. Enjoy!
Source: http://nerdvittles.com/?p=707
Magento Preview Version 1.4.2.0-RC1 – Now Available!
We are happy to announce the availability of Magento Preview Version 1.4.2.0-RC1 for download.
As this is a preview version it is NOT recommended in any way to be used in a production environment (more information about preview releases and the new community edition release process can be found in this blog post). We also highly encourage extension developers to test their extensions for compatibility with this version. This release is NOT available for upgrade through Magento Connect Manager and is only available on our download page.
This release includes the new Magento Connect Manager (MCM) and allows the community to start testing the new MCM and Magento Connect 2.
To see how to work with new MCM please visit our release notes page. Diff files are available here.
Please report all issues with this release in the bug tracker.
Source: http://www.magentocommerce.com/blog/comments/magento-preview-version-1420-rc1-now-available/
OSTAG Is Officially a 501(c)(3) Non-profit Corporation
OSTAG is pleased to announce that we have officially received word that we are now a 501(c)(3) non-profit corporation under US law. A special thanks goes out to Melanie Clark for her amazing efforts in this process. In addition to filling out numerous forms, and replying over and over again when the IRS came back with questions, she also met with IRS representatives on more than one occasion. Without her hard work and experience in such matters we would not have made it. Thank you, Melanie!
Now that we have achieved non-profit status we will begin our fund-raising efforts in earnest. For those wishing to make a donation feel free to use our Paypal account: donations@ostag.org. You may also contact us if you need to discuss alternative methods of payments.
Where will the funds go? Our number one goal is to fund open source telephony projects that can be licensed liberally and used for the public good. We are soliciting ideas for such projects, so please use the Contact Us link to submit your ideas and questions.
Looking forward to serving the open source telephony community,
OSTAG
Source: http://ostag.org/node/6
Use FreeSWITCH To *Receive* Free Calls With Google Voice!
Yesterday we talked about how Anthony added the ability to use mod_dingaling to make outbound calls using the new gmail voice interface. After taking a day off from doing ridiculously cool things with mod_dingaling, Anthony has now programmed the ability to receive calls in FreeSWITCH from a Google Voice account! This now means that you don't need to have a Gizmo number (or a landline) in order to receive incoming Google Voice phone calls into your FreeSWITCH system.
I've updated the GV call wiki page with a new section that discusses how to configure incoming calls. This is all brand new, so try it out and report back! Let us know how you're using your newfound GV features with FreeSWITCH.
-Michael
Source: http://freeswitch.org/node/281