ssokol February 18th, 2008

After nearly a year of cooking, AsteriskNOW! version 1.0 is ready for your dining pleasure. (So it took a year…. Blame the folks at Google. They’re the ones who made long-term beta tests popular. Some crazy idea about getting things right before they actually release them as products. Silly people.)

So, you ask, what is AsteriskNOW!? It’s a complete installation of Asterisk that doesn’t require a masters degree in Linux geekery to use. Download the image, burn it to CD, drop it in the tray, fire up the PC and 15 minutes later you’ve successfully converted an ordinary computer into an amazing telephony server with an easy-to-use web-based graphical user interface (a.k.a. the Asterisk GUI).

So, you ask, what can I do with AsteriskNOW!? Well, !?#& near anything. At least !?#& near anything that involves telephony. You can build a basic IP-PBX. You can VoIP-enable a legacy PBX. You can build a conference bridge. You can replace your ancient and limited voicemail system with a state-of-the-art Asterisk-based unified messaging system. You can make your one-man shop sound like a Fortune 500 enterprise with automated attendants (a.k.a. IVR menus). Oh, and you can do all of that from the comfort of the web GUI.

Want to take it up a notch? Ok. Dig in just a bit and you can build you own IVR system using the Asterisk Dialplan scripting language and either web-service or ODBC data sources. Want to stretch things a bit further? Learn the ways of AGI (the Asterisk Gateway Interface) and build a sophisticated voice communications application in the programming or scripting language of your choice. Feel like jumping in with both feet? Bust out your C compiler and build your own low-level applications using the Asterisk C API.

So what’s new since the last beta release? Great question:

• Over 1000 updates and improvements to the GUI
• Updated to Asterisk release 1.4.17
• Updated to Zaptel release 1.4.8
• Updated Linux Kernel 2.6.22
• Polycom phone auto-provisioning
• Improved package management and update capabilities

But wait…. That’s not all. This 1.0.1 version includes a few extras that aren’t part of the canonical Asterisk 1.4 release (including the Open Settlement Protocol and the phone provisioning module). Starting with the next release, AsteriskNOW! will be available in two flavors: AsteriskNOW! Pure, which contains only pure GPL and GPL compatible components and runs only the stock version of Asterisk, and AsteriskNOW! Plus, which includes non-GPL components and a limited number of features back-ported from the development version. As they say in in the burger commercials, have it your way!

One last thing: <shameless-commercial-plug>AsteriskNOW! is a great way for developers, VARs and ISVs to get started with Asterisk. Once you’ve seen the power, flexibility and feature set that makes Asterisk the most popular telephony SDK in the world, you’ll probably want to take a look at Asterisk Business Edition (ABE), which is AsteriskNOW!’s commercial counterpart. ABE uses the same simple installation, the same GUI framework and includes a range of support options and a flexible license model that make it ideal for commercial deployments.</shameless-commercial-plug>.

• AsteriskNOW , Community , Genuine Asterisk
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malcolmd February 18th, 2008

Howdy,

Many of you may fondly remember my last posting about AA50 software updates.

It’s been nearly 3 months, so how about another update?

We’re pleased to announce the release of the 1.1 software for our AA50.  Customers may download the software update from the Asterisk Business Edition downloads portal using their Digium.com account.

What’s new?

A slew of International-related analog trunk and station options have been exposed via the GUI in a new “Operation Mode” section of the Networking tab, in the Advanced tab of the Analog Service Provider configuration, and in the Users tab as a part of user creation. 

You can now set:

• Busy Detection on or off
• A numerical value for Busy Count
• Busy Pattern
• Ring Timeout
• Answer on Polarity switch and Hangup on Polarity switch
• Call Progress (still not super-reliable) and Progress Zone
• To enabled Caller ID, or not to enable Caller ID, when to start it (ring or polarity), whether you want your Caller ID as received or custom, and your CID signalling type
• Trunk and Station Flash timing
• Pulse or DTMF dial
• Country specific Opermode - for setting impedances on your trunks
• a-law or u-law codec for your trunks
• Whether to apply Opermode to FXO only or FXS and FXO
• Normal or boosted station ringer voltage
• Low power mode
• Fast Ringer mode
• Ring Detection - Standard or Full Wave  (to improve UK CID detection)
• Tone Zone - Confuse your friends and impress your grandparents by using “us-old”
• Signalling - Loopstart or Kewl Start (Loopstart with Disconect Detection)

Wow..that’s a lot.

But what about things other than analog trunk configuration?

G.722 transcoding support has been added.  Polycom 650 and 550 users rejoice.

A new mode of DHCP operation has been added to allow for the provisioning of Polycom phones on the WAN side of the AA50.  How does this work?  If you’ve got an existing DHCP server in your network, and it’s not already serving Option 66, then enabling this option will cause the AA50 to listen for DHCP requests, but not serve DHCP addresses, and then offer Option 66 to devices that are looking for it.

By default, there’s no twist in the transmitted DTMF values out the PSTN ports.  Now, we’ve got twist for Brazil.  Other countries, coming soon.

Date and Time may now be set manually via the GUI - so an NTP server isn’t required.

And, the Backups tab now allows you to download a local copy of a Backup, or for you to upload a Backup onto the AA50 - those of you looking to roll the same configuration to more than one unit, take note.

A synchronization between the Embedded Business Edition software on the AA50 and the Business Edition software that comprises the C1.5 release.

And, some improved (performance-wise) math for mixing channels in MeetMe conferences.

What’s fixed? 

The setting of static WAN IPs now works properly.  Yay.

Calling Rules no longer get stuck in an un-deletable (yes, that’s totally a word) state.

Some help text for Queues has been cleaned up.

Clicking on the Timezone sub tab no longer makes the WAN and LAN tabs disappear.

In User creation, the codec list now shows up by default.

An issue with sticky inbound call routes has been addressed - old deleted routes are now really deleted.

We’ve added some Polarity detection fixes to improve Caller ID and disconect detection.

A crash issue that surfaced while writing voicemails has been resolved.

In some cases, on first boot, the AA50 would hang while trying to generate an SSH key.  This no longer happens.

The Polycom timezone GMT offset has been re-fixed.  This time…it’s for real.

Special Thanks To:

Our customers, for pointing out things we needed to fix.

Our international partners, for helping us provide the necessary configuration tools for their locales.

The Academy, The Hollywood Foreign Press, The Golden Raspberry, my wife, our tireless engineering department, the Aromas Coffee Shop for opening a branch office inside of the Digium building and for keeping us fueled, and Mr. T as B.A. Baracus.

• Products
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beelinebill February 11th, 2008

Digium is putting our money where our mouth is. We are investing in total quality programs throughout the company - and today we are rolling out the new Digium Exceptional Satisfaction Program (ESP).

ESP includes:

• Quality hardware products with “Stand behind the product” warranties.
• Money back satisfaction guarantee.
• Courteous and helpful service agents.

The quality mantra starts with our customers and extends to every aspect of our products and customer service. Our goal is to produce the highest quality hardware and software and to deliver the highest quality business solutions, training and support to totally satisfy our customers. Over the past year, we have re-architected nearly the entire range of PCI Telephony interface and gateway cards and introduced a wide range of PCI Express cards. We have listened to the community and our customers and now offer an echo-free guarantee. We have refined the drivers and increased performance for these cards.

Effective immediately, on all current Digium cards we have set the PCI and PCI Express Card warranty at FIVE YEARS and offer a no-risk guarantee to our customers. (See our End of Life announcements for discontinued products.) We have improved our pricing on a number of products - and redesigned our channel programs to benefit those organizations that truly partner with Digium.

Digium PCI and PCI Express cards are the best value in the market. If the cards do not work with Genuine Asterisk as advertised, our top notch technical support team will work with you to resolve the problem. If the problem can not be successfully resolved, we will refund your money. Yes, you heard it. We have always done this, but now we are promoting it and will continue to. Digium’s Hardware Appliances offer standard 1 year warranty that can be extended by renewing the subscription on your product.

100% Customer Satisfaction. Make no mistake - Digium is dead serious. As benevolent sponsor and maintainer of the open source Asterisk project, Mark Spencer’s dream has forever altered the world of communications.

- - - - - - -

What do our customers say? Here is an excerpt from one of the open source Asterisk mailing lists, one person responding to another:

I have been using 220B’s for about 6 months. I have about 20 of them out in the field. I have not had any issues with them, and feedback is positive.

Same here. I’ve been using five TE220B in my company at 5 different sites since october 2007; up to now, zero problems and no echo at all. One of the sites runs a small callcenter that handles about 1000 incoming calls per day. So far the feedback is really positive. Alberto.

- - - - - - - -

I have launched many programs over the years, and while we’ve been planning this program and launch I have received more exciting employee feedback than ever before. The hallways, coffee pot, conference rooms, and parking lot discussions are about the excitement of proving to the world that open source telephony solutions are ready for the mainstream and so is Digium! We are proud of our “Mark Spencer” heritage as the author and creator of Asterisk as well as today’s major sponsor and community steward. We have the worlds best IP PBX in Asterisk, according to Infoworld, who awarded us the “Best IP PBX” award last month to kick off 2008. World class service and people along with the best open source based IP Telephony products on the planet.

Customer Focus! Quality Products! 100% Customer Satisfaction!

Visit www.digium.com for all the details.

We look forward to serving you!

• ABE , AsteriskNOW , Community , Genuine Asterisk , Open Source , Products , Quality
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malcolmd February 4th, 2008

Things overheard on the Interwebs:

“I wish Digium had a four-port analog card with hardware echo cancellation.”

“I wish Digium had a four-port version of their Voicebus analog cards.”

“I wish Digium had a common driver interface instead of so many different drivers.”

In the words of Robin Williams as The Genie, Digium responds:

“Excuse me? Are you lookin’ at me? Did you rub my lamp? Did you wake me up? Did you bring me here? And all of a sudden you’re walking out on me? I don’t think so, not right now. You’re getting your wishes, so sit down!

Ladies and gentlemen, boys and girls, geeks and nerds, Digium is pleased to announce the immediate availability of the TDM410.

The TDM410 is our new four-port analog interface gateway card and is our successor to the wildly popular TDM400 product. Improving upon the TDM400, the TDM410 utilizes our patent-pending VoiceBus interface, as already found on our twenty-four and eight-port analog interface cards, as well as our single port T1/E1 interface cards. With the use of this interface, Digium now maintains a common PCI and driver interface across the range of our analog products; reducing system incompatibilities and driver complexity and increasing ease of use.

For those customers that demand superior voice quality, the TDM410 supports a Digium hardware echo cancellation module.

For those customers that are already buying Digium’s existing single channel trunk and station modules; no worries, they’re 100% compatible with the TDM410.

The TDM410 is priced, without modules at $175 MSRP. The corresponding echo cancellation module, the VPMADT032, is priced at $235 MSRP. Single channel trunk modules are priced at $43.95 MSRP. Single channel station modules are priced at $40.95 MSRP.

“Thank you for choosing “Magic [Digium] Carpet” for all your travel needs. Don’t stand until the rug has come to a complete stop. Thank you. Goodbye, now. Goodbye. Goodbye, thank you. Goodbye”

[updated] Fixed a bad URL.

• Products , Quality
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