Congress recently passed the 2009 American Recovery and Reinvestment Act. Did you happen to notice the incentives for businesses to invest in their phone systems?

We didn’t either.

Therefore, Digium® is launching its own worldwide Telecom Stimulus Program to help businesses
invest in their most important piece of office equipment…their phone system!

Here’s the scoop…

Buy a Switchvox® appliance with SMB software and at least 20 subscriptions between
March 1st and April 30th 2009 and receive up to a $1000 rebate!

See the details at these links:

• Stimulus Package Program Documentation
• Rebate Claim Form

This really is a great deal…check it out today!

Digium is pleased to announce the release of 1.3 software for the AA50.

Users may download the update directly from Digium’s AA50 downloads portal

Users who have registered their AA50 hardware and their subscription, as included with their AA50 purchase, at http://www.digium.com/register may visit the Asterisk Business Edition download portal at https://be.digium.com. Once logged in, you will see a “Downloads” link. Click it. Then, you’ll see a tab called “AA50 Files.” Click it. You will now see a 1.3 Folder. Go there. Inside of this folder, you will see a Changelog file that contains information about what’s new in the release and the new firmware itself - a uImage file. You may download this uImage file to your local PC.

Once you have downloaded the uImage file, reboot your AA50. After it has finished, log into the web UI for your AA50. Once there, go to the “Updates” menu. There, you will see a form that allows you to upload to the AA50 the uImage file that you’ve downloaded to your local PC. Once you’ve uploaded the uImage file, your AA50 will need to reboot. The upgrade process can take about 8 minutes, so go grab a cup of coffee. When you come back from your coffee, log into the web interface again and the upgrade process will be complete.

AA50 Software Release 1.3

The new AA50 1.3 software release includes a number of enhancements and bug fixes, but is primarily a feature release that provides significant new features spanning multiple areas. Key new features provided include: G.729 support, Paging and Intercom, Findme/Followme, Custom Feature Codes, a new IVR application, a new real-time status page to see who’s on the phone and how much voicemail they have, the ability to do Bulk Adds, the return of the Active Channels page and the ability to manipulate VoIP phone digit mapping and digit timeout.

Additional New Features:

• Registration tool support for registering g729 licenses has been added to the GUI.
• Ability to upgrade Polycom firmware through GUI.
• Added the ability to assign multiple line keys to a user on a Polycom phone.
• Added digit mapping option for Polycom phones.
• Added Backup Options - A check box has been added in the AA50 GUI. Backups now have the option of including voicemails and custom prompts.
• Added ssmtp authentication option for emailing voicemails.
• Added the ability to view DHCP assignments\leases in the GUI, to assist administrators with managing phones, etc.
• Added the ‘tcpdump’ utility, accessible via the CLI.
• Added DTMF Detection via Features.configuration.

Enhancements

• Added the ability to select format in which to record custom prompts.
• Parked calls and conference calls are displayed on the “System Status” page of the GUI.
• Added an ‘Update Timezones’ button to the GUI

Other Stuff

In addition to the changes listed above, we’ve done a number of other things:

• Upload functionality now works for IE6 and IE7.
• Corrected erroneous “unregistered” display of IAX2 trunk on AA50 GUI status page.
• Corrected an issue where it was possible for a corrupt filename to occur when uploading a backup file into the AA50 using Internet Explorer.
• Corrected issue where calling inbound to an AA50 extension produces an inconsistent ring-back.
• Corrected display of IAX2 trunk on AA50 GUI status page

We get many feature requests from customers every day, and we get reports of bugs, too. We’ve done our best to address some of those requests and issues that we think will have the largest impact on the customer base. Your feature requests and issue reporting are important to us and we’re always interested to hear your suggestions.

Thank you and I look forward to speaking to all of you again in the not so distant future.

Well, it has been nearly six months since the announcement of Skype For Asterisk at AstriCon, so it is high time for an update.  So far we’ve been through several rounds of code with a very small pilot group.  The results are encouraging – since that initial demo call on stage at AstriCon, we’ve logged tens of thousands of hours of Skype-To-Asterisk communication.  We’ve also learned a lot about the art of connecting Asterisk with the Skype global network.  We continue to see a great deal of interest in this product (over 3,000 applications and counting) and we appreciate the patience of those waiting to test and ultimately consider purchasing it.  Hang in there, please!

For those of you who aren’t already in the know, Skype For Asterisk (SFA) is an add-on channel driver that integrates Skype calling with Asterisk-based telephony systems.  With SFA, businesses can build a presence on the Skype network, allowing customers to call in for free using Skype.  SFA also gives users business users access to low rates for inbound and outbound calling using SkypeIn and SkypeOut.

We receive many questions about what have Skype and Digium been doing since the announcement. The first step was to deploy the initial round of beta code for testing with a small group. That is now complete, and just this week we have opened up the testing to a second, larger group of testers. Once round two is complete the goal is to release a public beta and then the finished product. Beta feedback has been extremely positive so far and we have no doubt SFA will live up to its AstriCon hype!

The SFA product will be the only solution that integrates Asterisk directly with Skype. This is not a “proxy” solution and the call quality will be superior to anything else on the market. Customers will have the ability to make, receive and transfer Skype calls from within Asterisk phone systems, using existing hardware and existing Asterisk configurations: Skype calls become just another Asterisk call.

Some of the features that will be supported in the market release are:

1.       SkypeIn:  Receive calls from the public telephone network using standard phone numbers

2.       SkypeOut:  Make calls to landline and mobile numbers at incredibly low rates

3.       Standard phone features: Incoming/outgoing digits (DTMF), Caller ID

4.       Smart call routing based on called Skype Name, Caller ID, country of the caller, language they have chosen in their Skype client and etc.

5.       Retrieve Skype credit balance information

6.       Store and call PSTN and Skype contacts

7.       Retrieve and set Skype user presence information

8.       Support for G.711 and G.729 voice codecs

9.       Each Skype channel license includes a Digium G.729 codec license

Skype and Digium are working hard to deliver a first class solution that will allow businesses to save money, which is particularly important during the current economic times. We are striving to create a feature rich, stable solution with the best voice quality on the market today. 

We are confident that this goal will be reached in the very near future.  Please stay tuned to this blog for more updates and keep an eye out for the announcement of the public beta. 

Skype For Asterisk: Coming soon to an Asterisk near you.

Asterisk is becoming more and more a choice for larger installations, both in Enterprise as well as Carrier environments.  While SMB (small/medium business) continues to be the heart of the user base, it seems that many of the questions on the minds of the user and developer community have to do with scale, redundancy, and configuration issues which are relevant to larger installations.

To encourage the improvement and testing of larger-scale Asterisk systems, I’d like to repeat here what I mentioned today on the asterisk-dev mailing list:  I’m putting out a semi-official challenge in place.  The first person to get an Asterisk system moving 10,000 G.711 call legs through a single instance on a single machine will get a first-class steak dinner at Astricon.  And a great bottle of wine, if that is your preference.

This isn’t an X-prize, but the concept is the same - think of it as an S-prize.  ”S” means “Steak”.  Or maybe “Salad” if you’re a vegetarian. 

There are some serious hurdles here, for both software and hardware.  Ten thousand channels sounds like a lot, and it is.  But it can be done, and is already done with custom hardware from closed-source vendors.  Open Source Asterisk has not been (to my knowledge) tested at anywhere near that capacity, though attempts have been made in the thousands of channel level ranges with good successes.  However, there are significant “walls” to climb between ~2500 channels and 10000 channels, and this is not merely a linear application of processing power.  Just the throughput for this 10k challenge is pretty impressive: 500k packets per second per direction at 20ms, and 820mbps per direction.  Ethernet trunking/bonding may be required to overcome IRQ issues, or certainly very close administration of a gigabit interface.  Operating system tuning will be required in conjunction with Asterisk tuning (and probably patching.)  Documentation on how you achieve this will help everyone, and this is an “open” challenge.

I know that the Asterisk community is already headed in the direction of making these kinds of advances.  This is merely an incentive to speed up the process, and hopefully get some discussion going as to how people might solve the problems both in the Asterisk implementation as well as on the systems which are required to handle that kind of throughput.  Everyone benefits from the effort, and I think it can be done with some concentration and clever design.

Will there be some other platform that makes it there first, or has someone already reached that number?  Maybe!  I think any work done towards improving VoIP RTP throughput on Linux systems at these high packet-per-second rates will benefit everyone, especially if it’s an Open-Source solution that reaches the goal first.

Small print: one winner, even if a group effort - choose amongst yourselves. Group members may get secondary prizes.  Prize dinner will be at the following Astricon, in whatever city that is holds Astricon that year. Equipment must be “off-the-shelf” gear.  Software must be Asterisk from SVN, or patched with code that has been submitted to the bugtracker with a valid submission agreement for inclusion into Asterisk.  MOS of audio channels must be at least 3.2 or better (subjective.)  Calls must be standard G.711 with 20ms or 10ms sample intervals.  All RTP (RX,TX) for all channels must be routed through a single instance of Asterisk.  Methods must be documented and reproducible by other community members or Digium with the same equipment.  Winner will be publicly cheered.  Dinner to be at Mortons, Ruth’s Chris, or comparable venue.  10,000 call legs is equivalent to 5,000 “hairpin” calls, though 10,000 calls to “Echo()” would be acceptable.  Substitutions are possible.  SIP, IAX2, or H.323 are acceptable protocols.

Asterisk is becoming more and more a choice for larger installations, both in Enterprise as well as Carrier environments.  While SMB (small/medium business) continues to be the heart of the user base, it seems that many of the questions on the minds of the user and developer community have to do with scale, redundancy, and configuration issues which are relevant to larger installations.

To encourage the improvement and testing of larger-scale Asterisk systems, I’d like to repeat here what I mentioned today on the asterisk-dev mailing list:  I’m putting out a semi-official challenge in place.  The first person to get an Asterisk system moving 10,000 G.711 call legs through a single instance on a single machine will get a first-class steak dinner at Astricon.  And a great bottle of wine, if that is your preference.

This isn’t an X-prize, but the concept is the same - think of it as an S-prize.  ”S” means “Steak”.  Or maybe “Salad” if you’re a vegetarian. 

There are some serious hurdles here, for both software and hardware.  Ten thousand channels sounds like a lot, and it is.  But it can be done, and is already done with custom hardware from closed-source vendors.  Open Source Asterisk has not been (to my knowledge) tested at anywhere near that capacity, though attempts have been made in the thousands of channel level ranges with good successes.  However, there are significant “walls” to climb between ~2500 channels and 10000 channels, and this is not merely a linear application of processing power.  Just the throughput for this 10k challenge is pretty impressive: 500k packets per second per direction at 20ms, and 820mbps per direction.  Ethernet trunking/bonding may be required to overcome IRQ issues, or certainly very close administration of a gigabit interface.  Operating system tuning will be required in conjunction with Asterisk tuning (and probably patching.)  Documentation on how you achieve this will help everyone, and this is an “open” challenge.

I know that the Asterisk community is already headed in the direction of making these kinds of advances.  This is merely an incentive to speed up the process, and hopefully get some discussion going as to how people might solve the problems both in the Asterisk implementation as well as on the systems which are required to handle that kind of throughput.  Everyone benefits from the effort, and I think it can be done with some concentration and clever design.

Will there be some other platform that makes it there first, or has someone already reached that number?  Maybe!  I think any work done towards improving VoIP RTP throughput on Linux systems at these high packet-per-second rates will benefit everyone, especially if it’s an Open-Source solution that reaches the goal first.

Small print: one winner, even if a group effort - choose amongst yourselves. Group members may get secondary prizes.  Prize dinner will be at the following Astricon, in whatever city that is holds Astricon that year. Equipment must be “off-the-shelf” gear.  Software must be Asterisk from SVN, or patched with code that has been submitted to the bugtracker with a valid submission agreement for inclusion into Asterisk.  MOS of audio channels must be at least 3.2 or better (subjective.)  Calls must be standard G.711 with 20ms or 10ms sample intervals.  All RTP (RX,TX) for all channels must be routed through a single instance of Asterisk.  Methods must be documented and reproducible by other community members or Digium with the same equipment.  Winner will be publicly cheered.  Dinner to be at Mortons, Ruth’s Chris, or comparable venue.  10,000 call legs is equivalent to 5,000 “hairpin” calls, though 10,000 calls to “Echo()” would be acceptable.  Substitutions are possible.  SIP, IAX2, or H.323 are acceptable protocols.

Howdy,

Long time no blog, for me anyway. It’s nice to speak to everyone again.

I’m blogging, because today I am a solicitor - not in the legal sense, thank you. I’m soliciting testers for a new Asterisk fax product. This fax product improves upon existing Asterisk fax products by offering better multi-page performance with fewer failures, better compatibility with older fax machines, and lower bit-error rates in noisy environments.

Qualifications:

1. High Degree of familiarity with Asterisk dialplan construction.
2. Time to test. We need people that can hop to it, and test quickly.
3. Ability to devote a server and some Digium boards (supplied by you and already at your disposal) and / or fax-enabled VoIP service provider accounts (also supplied by you) to the testing.
4. A good usage case for using fax with Asterisk.

Interested parties may e-mail me directly, responding to each of the four above qualifications. That’s my first name, my last initial, at digium.com -that’s malcolmdREMOVEthisPART@digium.com.

Parties accepted into the beta and who provide useful assistance and feedback will be compensated with a free license of the Asterisk fax product upon its release.

Thank you, and I hope to speak to all of you again shortly.

We are pleased to announce a major new release of the Digium G.729 codec module for Asterisk. This release incorporates a number of changes, and in addition, includes a new performance testing tool which will make it easier for users to choose the best ‘flavor’ of the module for the particular system they are installing it on.

The changes are:

- Version Numbering

The module’s version numbering is now more in line with our other products; the numbers will be in the form ‘X.Y.Z_A.B.C’, with the ‘X.Y.Z’ component representing the version of Asterisk the module is intended to operate in, and the ‘A.B.C’ component representing the actual version of the codec module code itself. Please note that the 1.6.0 modules will work in any Asterisk 1.6 release from 1.6.0 onwards, until such time as a new version is needed; when that occurs, we will provide 1.6.0 modules *and* modules designed for the later releases.

- Optimization

The new modules were built using the latest (4.3) release of the GNU C compiler, and use a number of new optimization methods available in that release to significantly improve encoding/decoding performance. In addition, we now build the modules in both 32-bit and 64-bit varieties for each CPU flavor that support both modes, so that users with 64-bit CPUs running a 32-bit Linux installation can still have a version of the module optimized for their CPU type. GCC 4.3 also provides optimizations for some newer CPU families (Intel Core2, AMD Barcelona, and others), so we’ve made those flavors available with this release.

- Performance Testing

In the past, we’ve offered different CPU flavors of the module to hopefully provide the best encoding/decoding performance, but it was up to the user to install and test each flavor to determine which one provided the best performance. This process is difficult and time consuming, and did not always provide reliable results. To help with this situation, we’ve now released a tool called ‘benchg729′, which can be run on the target system and will execute encoding speed tests (using a real audio sample) for each CPU flavor that we offer for that platform, and then report the results and suggest the best performing module flavor for that system. Note that the use of this tool requires that the system have at least one valid channel license for the Digium
G.729 codec installed; it will not run without a license.

The new codec modules and benchg729 tool are now available at:

http://downloads.digium.com/pub/telephony/codec_g729

New codec modules are available for Asterisk 1.2, Asterisk 1.4 and Asterisk 1.6.x on both x86-32 and x86-64 Linux platforms, and the benchg729 tool is also available for x86-32 and x86-64 Linux platforms.

We hope these product updates improve your system performance, and as always, we thank you for supporting Asterisk and using Digium products!

Being at the right place at the right time can either be the result of visionary foresight and meticulous planning – or a fortuitous turn of events. If you live in Miami, then you are the beneficiary of a fortuitous turn of events given that IT Expo/ Digium Asterisk World is being held in your fair city beginning next Monday. For the rest of us, given that half the country is currently being blanketed by ice and snow, a little meticulous planning to make our way to sunny south Florida for the show seems very worthwhile. Plus, who could object to a brief reprieve from winter?

IT Expo/ Digium Asterisk World is the perfect venue to better understand why open source telephony is a technology that is at the right place at the right time. Underlying weakness in the economy is forcing end-users to conservatively allocate each and every dollar spent. Voice over IP is a technology that offers a demonstratable return on investment and is predicted to surpass traditional telephony in the number of lines installed this year. Couple the movement to VoIP with the increasing adoption of open source as a lower cost alternative to traditional proprietary solutions – and you get Asterisk.

We’d like to think that Asterisk emerged as the world’s leading open source telephony platform due to visionary foresight and meticulous planning; however, it’s really more of a fortuitous turn of events. Who, ten years ago at the time Asterisk was begun, would have predicted today’s simultaneous intersection of high level trends that are now driving the adoption of Asterisk - namely, VoIP, open source, and a struggling economy? As a result, the numbers associated with Asterisk adoption are truly amazing. I’ll be talking about these Asterisk adoption statistics along with more detailed information on the industry and economic trends mentioned here during my keynote address at IT Expo/Digium Asterisk World. Please join me at 10:30 on Tuesday in room B217/218 for the address. Hope to see you then – and don’t forget to pack your flip flops!

Many years ago, I accepted a sales position which required me to relocate from Chicago IL to Dallas TX. As a frequent traveler, I had a relationship with a trusted travel agent in Chicago who I continued to use when I relocated to Dallas. I often flew on American Airlines, and my travel agent helped me choose the right American flights to keep me on time and within budget. A few months into the new job, my company decided that all travel would be coordinated through a new travel agency who was selected to reduce costs. The new travel agent began to book me on an upstart “alternative” carrier called Southwest Airlines. Yes, I’m old enough to remember Southwest as an upstart! Southwest got me to the places I needed to go, on time, and significantly reduced my air travel expenses. I was a happy camper, as was my company.

My former travel agent called me and asked why I had not been booking flights with her. I explained the situation and mentioned that I had started flying Southwest. She indicated that she could book me on Southwest if I would move my business back to her agency. This offer begged a question. I asked “Why didn’t you tell me about Southwest Airlines 3 months ago?” Her answer was “Well, you never asked”.

There are a couple of lessons to be learned from this story. First, if your customer has to ask you for the specific solutions to help their business, are you really their trusted advisor? Second, if you don’t tell them about value solutions, someone else will.

With sobering economic news bombarding us each day, it’s important to understand how to position your company to leverage reductions in IT spending to your advantage. Here at Digium, we have seen demand grow significantly for both our Switchvox and open-source Asterisk solutions as customers look for solutions which offer more value to their business. Economic realities are forcing customers to look at “alternative” brands to complete projects that began when only “incumbent” brands were considered as viable options.

As a Digium reseller, you are well-positioned to take advantage of the current environment by showing your customers how they can reduce operating costs, increase revenue, and improve employee productivity by investing in their voice communication systems. Digium’s solutions can accomplish all this, and provide a much faster return on investment than traditional solutions. As trusted advisors to your customers, you are in a unique position to introduce them to “alternative” brands at a time when they are most willing to listen.

If you are a reseller or integrator looking to win more business in this tough economic climate, come visit us at Digium Asterisk World!

http://www.tmcnet.com/voip/conference/east-09/digium-asterisk-world.htm

Jim Butler

Digium, Inc. | Director of Global Channel Sales

445 Jan Davis Drive NW – Huntsville, AL 35806 – USA

256-428-6019 - 1 number finds me!

jbutler@digium.com

If you’re planning to travel, you can usually find a guidebook tailored for your specific interests- Hawaii for cyclists, Panama for foodies, and China for art lovers. Well, think of this post as your guide to Digium Asterisk World for Switchvoxers…er Switchvoxites, Switchvoxists?….Well, you get the idea- if you’re looking to learn all you can about Digium’s award winning line of Switchvox phone systems for small and medium businesses, here are the sessions and events you’ll definitely not want to miss!

1. Take a Switchvox Training Course - Tuesday, Feb 3, from 8:30am-5:00pm with a break for lunch, you can learn everything you need to get started with the entire line of Switchvox PBX systems. By the end of the day, you’ll know all about what Switchvox is, how to prepare to install one, the steps needed for physical installation and you’ll have a handle on the features that companies use most, including Auto-Attendants, Voicemail and Conferencing. Added bonus: Every attendee to the training course gets a Digium TDM411B Analog card (1 FXO port + 1 FXS port) and a Polycom SoundPoint® IP 330 SIP Phone along with a t-shirt, bag, stickers and other nice schwag. Sweet!
2. “Web-Aware Unified Communications with Switchvox” – On Wednesday, Feb 4 from 8:30-9:15am come join me for the first session on the Reseller conference track and learn how mashing up Unified Communications features and web applications allows your customers to make better, faster decisions at work, and get business done. If you’d like to see the future of phone systems, don’t miss this one! And I’m not just saying that because it’s my session.
3. “Ingredients for Successful Asterisk PBX Sales” – On Wednesday, Feb 4 from 1:30-2:15pm I’m moderating this panel where Switchvox is sure to be discussed a fair amount. I’m lucky to have joining me as panelists some of the best and brightest in the industry and to be perfectly honest, some of my favorite people I’ve met through shows like Digium Asterisk World. Ivan Kohler of Freeside, Bryan Johns of Shelton Johns, Peter Weyant of Interlink, Ravi Sakaria of VoicePulse and Michael Munger of High Powered Help will all be on hand to describe the features most requested by customers, the strategies they use for comparing against big iron systems, and how resellers can leverage these specific elements for successful advertising and sales of Asterisk-based systems. That’s going to be awesome, frankly.
4. Visit the Digium Booth for a Switchvox Demo – During every hour the exhibit hall is open, we’ll have 3 available stations where you can get a demo of a live system in the Digium booth, D 01. Getting a demo is a great way to get your questions answered about the systems specific capabilities- sure, you know Switchvox has got voicemail built in, but in the booth, you can see and hear how voicemail system really works. Little known fact: we won’t shoo you away if you want to avoid long distance charges and use the phones in the booth…as long as you tell your friend that the great sounding phone call is coming from a Switchvox PBX!

I really hope to see you at the show, it’s going to be the best Digium Asterisk World ever for anyone interested in Switchvox!

As hopefully everyone knows, there is an upcoming Digium/Asterisk World that will be at the ITEXPO show starting on February 2nd in Miami.  What will be happening?  Lots!  We have a great line-up of speakers across the three days.  With a single track each day, it’s a bit less intense than Astricon but has a wider range of offerings from the rest of the ITEXPO show and conference.

I’ve been helping with the line-up of speakers, and I think we’ve got some great topics on tap for the conference from both the business and technical camps.  Here’s a quick sample of some of the talks:

   - Nir Simionovich (GreenfieldTech) will be talking on virtualization of Asterisk for contact centers, which is a hot topic this year.  Perhaps he’ll have some additional details in his talk on the Amazon EC2 tests that he and Ronald Lewis have been doing, but certainly he’ll be speaking about some of the other non-cloud virtualization implementations he’s worked on for multi-seat contact centers.

  - Bryan Johns (Shelton|Johns) talks about the ROI on Asterisk systems, and how this allows resellers to make their case on the value of Asterisk-based platforms when comparing against proprietary platforms.

  - Our all-star panel on Monday will interact on the subject of “Lessons Learned” in the Enterprise/PBX environment, to instruct attendees on what to do when implementing Asterisk in a large environment (and probably what NOT to do, which is equally as important!)

There is a whole day dedicated to Resellers, to Enterprise/PBX users, and to Contact Centers.  No matter in what area your Asterisk interests are, there are bound to be talks that you won’t want to miss!

PS: The high temperature on Saturday in Miami is predicted to be 74F/23C.  As if the conference wasn’t attractive enough!

digium: Digium’s Telecom Stimulus Act http://tinyurl.com/b8kaxh

digium: Digium AA50, 1.3 Software Release http://tinyurl.com/c3oxo3

digium: RT @asteriskpbx: Release Summary for Asterisk Releases http://tinyurl.com/dfh4bw