Features that were previously only available through the Web interface are now made possible through Phone Feature Packs to phone handset displays, enabling users to access features from the phone handsets. Phone Feature Packs have replaced Phone Tokens in Switchvox SMB and now bring our most popular applications to snom and Polycom IP phones:
• Visual voicemail makes it easy to check your messages
• Searchable company directory lets you quickly find and dial co-workers
• Call recording from IP phones
• Phonebooks of commonly dialed numbers are accessible right from the phone
• Failover to a backup Switchvox
• Distintive Ring tones for different types of calls
• Support for multiple extensions on a single handset
• Hands free click-to-call from Switchboard, Notifier and Firedialer
• Additional security options to keep your VoIP system safe and running smoothly

Switchvox SMB 4.5 offers additional features to support users and administrators:
· User profiles— Caller profile information such as photo, extension, title and location. This appears on the Switchboard and on the display of phones with Phone Feature Packs on internal calls.
· Flexible language support— The Switchvox 4.5 GUI, manual, and inline help are available in English, UK English, Italian, Castilian Spanish, and Latin American Spanish. Sound packs, which include all the audio prompts used in the system, are available for these languages, as well as Australian English, French, and French Canadian.
· Comprehensive monitoring—Digium has implemented the Simple Network Management Protocol (SNMP) for Switchvox 4.5, which gives administrators the ability to collect real time data about the status and health of their systems.

SMB 4.5 is available now free of charge to customers with a current Switchvox SMB software subscription, and customers can continue using Switchvox 4.0 or implement the optional upgrade to 4.5 via Digium Website.

The ABP Team

Digium also makes and sells Switchvox, a turnkey unified communications system (IP PBX) based on Asterisk.  Switchvox is far less expensive than competitive IP PBX and UC systems based on proprietary technologies, but it is not free.

Users and customers frequently asked why Digium offers both the free-and-open Asterisk engine and the commercial Switchvox solution.  The answer is simple: while both products fit into the larger universe of telecommunications technologies, they have very different purposes and are geared towards very different audiences.

Asterisk is built by and for communication systems developers.  The open source project began in 1999 when Mark Spencer released the original Asterisk source code and began accepting submissions from a growing community of users.  The resulting product is an engine that handles all of the low-level details of initiating, maintaining and manipulating real-time media streams (calls) between endpoints (phones).  Since the initial release it’s been tested and refined by a community of more than 65,000 developers and integrators in 170 countries around the world.

Asterisk is to telephony what the Apache server is to web applications: essentially the exquisitely complex plumbing on which other applications are built.  Just as a web server does very little without web applications, a telephony server does nothing without telephony applications.  Web applications can be as simple as single static HTML page or as complex as Facebook or Google.  Likewise telephony applications can be very simple scripts or hugely complex suites of application software.

Low-level engines like Asterisk and Apache are extremely powerful precisely because they have no fixed function or specific purpose set by their creators.  The functions to which they are ultimately applied are determined not by the creators (the developers of the Asterisk and Apache development teams) but by application developers.

Application developers take engine-level components like Asterisk and Apache and build on top of them.  These developers craft purpose-built solutions that implement a specific set of functions.  Asterisk application developers write programs that make Asterisk behave as a PBX or as VoIP gateway or as a dialer or virtually an other type of telecom apparatus.

Some Asterisk applications are simple and use little more than the core Asterisk engine, a few configuration files and some scripts written in Asterisk’s Dialplan language.  More advanced Asterisk applications connect Asterisk with databases, web services and other external resources.  Finally, there are application suites that interconnect Asterisk with many other applications in a complex web of interactions.  These complex aggregate solutions do far more than could be done by Asterisk alone.  Digium’s Switchvox phone system is a perfect example of this class of application.

Where Asterisk is an engine, Switchvox is a complete vehicle.  The Switchvox development team has spent the past six years creating a powerful unified communications system that anyone with a minimum of computer experience can manage.  Where Asterisk is built for telecom developers, Switchvox is built for small and mid-sized businesses that need a powerful, cost effective phone system.

The Case For Switchvox

Digium’s line of Switchvox IP PBX systems make unified communications capabilities available to small and medium businesses.  Switchvox is administered through an easy to use graphical user interface (GUI) rather than raw configuration files and custom scripts.  Switchvox includes all of the standard features of phone system plus unified communication capabilities like advanced voice messaging, instant messaging, desktop fax, drag/drop call control, multi-party conferencing and advanced IVR.  Features that would cost thousands to bolt onto a traditional phone system.

With raw Asterisk, the process of configuring phones is entirely manual.  Each phone must be independently set up by the system administrator.  Switchvox automatically detects and configures phones, making it easy to deploy and manage users.  Switchvox also detects and configures Digium interface cards, making it easy to connect to the PSTN.  Setting up SIP trunks and tie-lines to other VoIP systems is even easier.

So who should really pick Switchvox instead of Asterisk?  People who aren’t telecom gurus who need a powerful, easy to install, easy to maintain, reasonably priced phone system for up to 400 users.

Don’t get me wrong: it’s entirely possible to create a powerful PBX system using raw Asterisk.  The major drawbacks to running Asterisk as a PBX are the deployment time and maintainability.  Building an IP PBX out of raw Asterisk requires some fairly advanced technical skills, including a good working knowledge of IP networking, Linux/Unix system administration, traditional telephony and script programming.  Even those who are fully versed in all four of these disciplines will need to overcome something of a learning curve to create a working system.  Once the system is up and running you will need someone on staff (or at least on call) who knows how the system works and how to handle any moves, adds or changes.

Digium’s flagship Switchvox SMB system (with all the bells and whistles you can imagine) starts at only $3600.  Our basic SOHO package is only $1600.  If you’re still tempted to use Asterisk, that’s cool but first do this:  Divide $1600 by what you think an hour of your time is worth.  Let’s use $50/hour as an example.  $1600 / $50 = 32 hours.  If you can learn enough Asterisk to build your own solution in 32 hours or less, go for it.  If not, take a good look at Switchvox.

The Case For Asterisk

Let’s go back to the engine/vehicle metaphor.  Asterisk is an engine.  It’s powerful.  It’s flexible.  It has enormous potential.  What it requires is a skilled engineer (or even a skilled shade-tree mechanic) who can take the engine and build it into a vehicle.  If you are creating a product or a custom solution that requires integrated voice communications, Asterisk is exactly what you need.

Let’s take the product scenario first.  If you want to build a conferencing server that connects to both VoIP and PSTN networks, Asterisk is a great starting point.  Asterisk has all kinds of features that make multi-party conferencing really, really easy.  It also includes native support for every major VoIP and PSTN protocol in use today.  To build a conferencing server out of Asterisk you need to pick out your platform hardware (computer), create an administration interface (probably a web application running on Apache) and possibly an end-user interface.  You’ll probably want to integrate with calendaring systems like Exchange, iCal, Google Calendar, etc.  You probably want to tie in email and possibly IM notifications and reminders.  Given a skilled development team you can probably bang this out in a few months.

Compare that with building from scratch and you can see the power of Asterisk.  You didn’t have to write (or license) a SIP stack.  You didn’t have to write your own DTMF detection algorithm or even wrap a DTMF collection function call for use in your application.  In fact, the actual “telephony programming” probably came down to a few dozen lines of Dialplan script and a bit of SQL to set up the database.  You shaved years off your development and testing path, added value through your snappy web interface and built it all on a free engine.  Nice.

Asterisk fits very nicely into the toolboxes of telephony integrators and data VARs.  If you’ve ever done custom integration work you know how difficult it can be to make systems from different vendors (or different generations) play nicely.  In enterprise scenarios where modern data applications share space in the server room with legacy switching gear, Asterisk can be indispensable.  It acts as a kind of “telephony glue” that ties VoIP to TDM and digital to analog.  It also bolts onto legacy systems as a perfect low-cost adjunct.  Your customer has an Octel voice messaging system that’s on its last legs?  No problem.  Replace it with an Asterisk-based system.  Your biggest client needs a dialer that can call an entire city in an hour?  Sure.  Asterisk can do that.

If you’re already familiar with networks, telephony and scripting, the Asterisk learning curve is fairly easy to overcome.  Read Asterisk: The Future of Telephony by Smith, Madsen and Van Meggelen.  Take a look at the samples and recipes on Asterisk.org.  Take the Asterisk Fast-Start or Asterisk Advanced class for a bit of hands-on training.  You’ll find that building solid solutions with Asterisk is drastically easier than building your own voice engine from scratch using a raw C language API from some proprietary vendor.  (Trust me on this one: I built IVR engines on the Dialogic and Brooktrout APIs before I discovered Asterisk.)

Asterisk is also a terrific way to learn about telephony and communications.  Students, hobbyists and artists have used Asterisk to build some extraordinarily creative applications while at the same time learning about telecommunications.  Some of the most successful developers in the Asterisk ecosystem started out experimenting with the code while in college or even high school.

Conclusion

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.

Call your ABP sales representative today at 972-831-1600 or email sales@abptech.com

Right now digital signage is as prevalent as ever. You can’t go out without seeing numerous video monitors displaying useful information or promotional material related to the business you are in. And obviously there is a great deal of demand for Signage services and the right products to meet your customers’ needs. These could be the ability to show graphic images or video at any type of retail/corporate/hospitality/financial/academic facility with video monitors scattered about. But what if you could also incorporate live video content from from sources like cable or satellite TV as well?

Introducing Cabletime products with the ability to distribute IPTV and IP Digital Signage content to any monitor you have via your network. With the 760/2M standard definition or 780/4G high definition models, you could display almost any type of content you want. And since these are IP endpoints, your customers would be able to manage a large number of displays remotely from their office. All management of the devices is through a web interface which easily lets users create playlists from multimedia content and control what each specific monitor displays at exact times of the day. Decoder boxes can cycle through pre-made content consisting of images, video files, PowerPoints as HTML pages and live video. At certain locations, it may be better to have a static image or fixed channel of live video constantly on some monitors. This is called ‘Standalone Mode’ and you could even change these “virtual channels” with a Cabletime IR remote control. All of this is possible with these IP Digital Signage and IPTV products from Cabletime. There are several models to choose from. The 760(SD) and 780(HD) both come in models with no internal memory that are purely IPTV receivers for live video over the network. The 760 has a 2GB version available now with options to increase up to 8GB upon request. The 780 ranges from 4-64GB of internal memory and there are also PoE options available as well. There is an unbelievable amount of control and customization with these small, yet powerful Decoder devices.

The German company MOBOTIX AG is known as the leading pioneer in network camera technology and its decentralized concept has made high-resolution video systems cost efficient. And the wireless Altai solution makes the deployment of Mobotix cameras simple and easy.

Announcing New Strategic Alliance, Patton and ABP Storm the Americas VoIP Market

Patton signs ABP Technology (www.abptech.com) as a master distributor for award-winning SmartNodeTM VoIP Gateways in the Americas as the partnership “Storms North America” with special pricing.

GAITHERSBURG, Md. and DALLAS, TEXAS — January 13, 2010 — Patton, the leader in business and carrier-class network access, connectivity, VoIP and triple-play equipment — and ABP Technology — value-added distributor of IP Telephony and IP Video solutions — have announced a strategic partnership that appoints ABP as a Master Distributor for Patton’s SmartNode™ VoIP Gateway product line in the Americas.

Patton has engaged ABP to distribute VoIP equipment to key reseller partners in conjunction with a blitz on the North American VoIP market that began in October.

During Q4 2009, Patton offered special year-end pricing on Patton’s two best-selling SmartNode™ VoIP gateway routers.

Although Patton’s promotions went out with 2009, ABP has extended the special pricing through the end of February 2010.

Resellers can view the ABP online partner storefront for Patton’s complete range of VoIP gateways that includes carrier-class products for service providers and enterprise-class products for small to medium-size businesses.

“As the telecom industry shifts to VoIP,” said Robert Messer, ABP’s President, “the market needs carrier-class products at cost-effective price levels. We chose Patton SmartNode because it offers the industry’s best price-performance for carrier and enterprise-class gateways. Of special interest to service providers is Patton’s touchless provisioning and configuration feature in SmartNode CPEs.”

As attested by Patton’s numerous RealityCheck® success stories, SmartNode combines the maturity, stability, and technical innovation that businesses and carriers demand.

“Patton equipment is manufactured in the USA,” Messer adds. “American manufacturing is a major benefit for our government sales. It also boosts our transactions with partners in NAFTA countries and Latin America.”

Manufactured at Patton’s Gaithersburg, MD facility, SmartNode™ recently claimed the 2009 INTERNET TELEPHONY® Excellence Award — the seventh major award in five years for the product line.

“ABP stands out as a specialty distributor in the IP world,” said Patton’s North America Sales Director, Sean Gerrity. “We are very pleased to partner with ABP to build strong channel-partner relationships throughout the Americas. ABP’s longstanding experience combined with strong market share in the service-provider and reseller sectors is especially attractive.”

ABP will showcase Patton’s award-winning SIP trunking solution — the SmartNode 4960 PRI VoIP IAD — at ITExpo in Miami during January 20–22 in booth 112.

Also, this firmware upgrade will allow for users to take advantage of the H.263/H.263+ video codecs that have been long awaited. This just allows for better interoperability with other products, specifically ABP’s Mobotix IP Surveillance Cameras. Now you can make (and receive) H.263 SIP Video calls from your Mobotix cameras to keep an eye on the security of your home or business. Configure your camera to call your phone if motion/noise is detected and be instantly aware of what is taking place.

Add these features to the Internet radio channels, the online photo galleries, and the powerful web browser and it is apparent that the Grandstream GXV3140 is a powerful multimedia IP video phone. It will be beneficial to those seeking a well rounded communications tool for business and personal purposes. And with such an aggressive introductory price, why wouldn’t you replace your audio only phones with one capable of face to face video calls and Wi-Fi connectivity? It makes sense to consider this when purchasing your next phone system from ABP.

Right out of the box upon the initial power up, the 3140 registers itself to the Grandstream IPVIDEOTALK network. You will be assigned a 7 digit number unique to your phone. This allows instantaneous, FREE video calls to take place between 3140’s which is ideal for personal use with family or those outside of your network. For corporate use, register your Grandstream to a Switchvox or another video capable IPPBX to make video calls with an extension just as you would usually do with old fashioned voice-only calls.

Or, keep an eye on the security of your business by making or receiving IP video calls with your Mobotix IP Cameras. Depending on your configuration, you could call the camera as an extension on your PBX or make a direct IP video call to whatever camera you would like to keep an eye on. Not enough for you? How about having your camera make a video call to you the second something is out of the ordinary. Utilize the features of your camera to trigger a call if there is a loud noise, motion, smoke detection and so on. If you are on the move, the same thing could be done using ABP’s Unidata WiFi video phone so that you are always aware of what is going on.

JUST AVAILABLE:
The WiFi USB dongle allows the 3140 to connect seamlessly to surrounding wireless networks working with the latest 802.11n standard as well as b/g. The wireless adapter can be plugged into the GXV-3140 via the USB port located on the right hand side of the phone for connection speeds of over 150Mbps. Now you won’t be restricted by wires or location with your Grandstream video phone!

These phones have excellent quality and a very aggressive price point. Call ABP today for more information and ordering of the Grandstream GXV-3140.

As a benefit of the American Recovery and Reinvestment Tax Act of 2009, beneficial provisions of IRC Section 168(k) (relating to bonus depreciation) and Section 179* (relating to increased deductions) have been extended for new equipment and software placed in service.

As a result, your small business customers don’t have to postpone investing in Cisco® Small Business technology products that will help them work more securely and effectively anytime, anywhere. That’s good for their business. And good for yours, too.

Help your customers take advantage of this tax break today. Tell them today but make sure to mention they need to check with their CPA if this will apply to them.* Bonus depreciation is available for all businesses and provides an immediate depreciation deduction of 50 percent of the cost of new equipment and software placed in service during calendar year 2009.

For taxable years beginning in 2009, 100 percent of new equipment and software purchases up to $250,000 can be expensed immediately.To limit this benefit to small businesses, the maximum immediate expense amount is reduced dollar-for-dollar to the extent that a taxpayer’s total new equipment and software purchases for the year exceed $800,000.

The following document details the configuration of a Mobotix M12 and a GrandStream GXV3000 allowing direct IP to IP calls delivering two way video and audio.

GrandStream GXV3000

The Mobotix and GrandStream can be configured to connect to a server that can handle the video codecs or can be configured to work directly with a compliant unit.

This document the method used to configure the two units as Directed IP calls, bypassing any server.

These instructions assumes a basic knowledge of the web interfaces for the related units. This documents also assumes your items are ready powered and ready to be configured.

Mobotix - M12

Web Interface à Audio and Phone à Phone Profiles

1. Give the profile a name - Location-m12

2. The “Phone number or SIP address” field is to be populated with SIP URL 100@10.3.48.167 ( EXT @ IP of Mobotix camera)

3. Specify the # of Dial Attempts and the Dial TimeOut

4. Choose SIP VIDEO for Connection Type

5. Message Type is the Greeting to be played once the call is established.

6. Confirm call with PIN code: (OPTIONAL) used to ensure the intended party receives the call

7. Choose “After the Message has been sent:”

A. Hangup - Do nothing

B. Listen - listen using the on-board microphone

C. Speak - Send audio to the on-board speaker

D. InterCom - two-way conversation

8. Click Set then Close to apply (but not save) settings

9. Once the profile has been applied, at the TOP of the page, you will see your profile name and a TEST button. Click TEST to check the parameters

10. Choose admin and Reboot camera

## NOTE ##

Clicking the Add button will allow configuration of multiple destinations to use this specific profile. Clicking the ADD PROFILE Button starts a additional NEW Profile.

GrandStream - GXV3000

Web Interface à Account X

The below only references fields that are modified, fields not mentioned require no additional configuration for operability

1. Choose Account to be configured

3. For Account Active - choose YES

4. For Account Name - enter a name

5. SIP Server - Enter the IP of the Mobotix Camera

6. SIP UserID: Input the extension configured within the Mobotix (see Step 2 in Mobotix section)

7. Name Field - Name only relevant to GrandStream Phone

8. UserID is Phone number - YES

9. SIP Registration - NO

10. VOICE CODEC’s Choice 1. PCMU = Choice 2 PCMA.

VIDEO - Choice H.263

We don’t normally re-post other articles and blogs but this short new piece by ChannelWeb is right on, and can hep ABPTech’s many non VoIP Market resellers that are deciding which Telephony platforms they want  to specialize in.

By Samara Lynn, ChannelWeb      11:31 AM EST Wed. Nov. 11, 2009