What is it?

The new Digium phones are High-Definition IP Phones designed by Digium specifically for Switchvox UC System and Asterisk-based phone systems. There are 3 phones in the series, the Digium D70, D50, and D40.

What does it do?

The Digium phones connect seemlessly to any Asterisk-based phone system, including Switchvox, Digium’s UC System.  When connected to an Asterisk Phone System, Digium Phones offer a user friendly calling experience complete with HD Voice Wideband and access to user created apps and call rules, directly from the phone. 

Being created by Digium specifically for Asterisk, Digium phones may very well have the easiest plug and play of any IP Phone. Simply plug the Digium phone into the network and the phone will automagically discover the Asterisk server, identify the phone user, and start communicating. 

Who is it for?

Digium phones are great for anyone using an Asterisk or Switchvox IP phone server.  For its initial launch, Digium released 3 models, the D70, D50, and D40 which suit different types of phone users. 

The Digium D70 being most feature rich is best suited for administrator or executives. This phone has 6 SIP lines, a 4.5in screen, and busy lamp indicators (BLF) for up to 100 contacts.  Perfect for someone who is watching employee phone time or doing many call transfers.  This model also is the only phone in the lineup with a dual Gigabit switched Ethernet ports.

The D50 is a 4-line phone with 10 rapid dial keys with BLF. With a 3.5in backlit LCD screen and dual 10/100 switched ports, this phone is perfect for mid-level or call center employees.

The D40 is a budget friendly 2-line IP Phone with a 3.5in backlit LCD. With its dual 10/100 switched PoE Ethernet ports, full duplex speakerphone, and 4 feature keys as well as access to Advanced Phone applications through the context sensitive softkeys, this phone is a perfect fit for just about any light to medium phone traffic office worker.

Availability?

So here is the ugly part…. Now that we are all hyped up and ready to try these phones out, they will not be available for shipping until April 2012. You can place your preorders now, though and as soon as we get some, we will send them out to you!

Want to try them all? VoIP Supply has created a handy dandy Lab Pack including one of each phone so you can sample each of the new Digium phones in your environment before you buy them in bulk.

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!

If you’ve been comparing hardware, are in the market for a resiliency solution, and in need of how-to resources for maintaining Asterisk based communications in the event of a hardware or software failure; you are now in luck.

Xorcom has taken has taken the time to offer a point-by-point comparison of their software and devices in an article of their own.

Xorcom TwinStar Comparison / Capabilities

Click here to read the Xorcom piece, Clarifications About Digium R-Series vs Xorcom TwinStar Comparison, that examines a Xorcom TwinStar plus Xorcom Astribank  solution that includes:

• Resiliency functionality across T1 Cards, Mixed Interfaces, and FXS Resiliency.
• Cost and effort to install the solution based on FXS and FXO port requirements.
• Relative cost and flexibility in the number of E1/T1 ports needed in a Xorcom solution.
• Hardware platform explanation.
• Functionality of TwinStar firmware.
• Digium R-Series compatibility issues.

Open Source Community

Xorcom, like VoIP Supply, supports building your own customized VoIP application and their article nicely sums up this hardware comparison and why they wanted to clarify Xorcom functionality:

Xorcom has always been a strong supporter of the Asterisk open source community and we believe it to be of the utmost importance to provide accurate information to the community.

Related Articles

• VoIP Failover with Xorcom TwinStar
• Digium R-Series Redundancy vs. Xorcom and RedFone
• Clarifications About Digium R-Series vs Xorcom TwinStar Comparison
• Xorcom Astribanks Explained
• A Guide to VoIP Gateways

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!

The newly released Digium R-Series redundancy appliances enable your network to recover from hardware or software failures and maintain communication with the outside world.  

Digium failover appliances work with their open source telephony platform Asterisk to ensure that you have an open line of communication even in demanding enviroments.

There are two models in the R-Series.  They work on Asterisk based open sourced PBX’s but do not work on Switchvox.  Both models are 1U rack mount devices:

• Digium R800:  Supports up to eight analog (POTS) circuits.
• Digium R850:  Handles up to eight digital (T1, E1, PRI or BRI) spans.

What’s The Difference?

So even in the event of a catastrophic failure, your communications system will stay up and running with the Digium R-Series.  That sounds great but, what makes these appliances different from other redundant systems that work with Asterisk?

With the help of Digium engineer Pete Engler, here’s how the R-Series stacks up agains other popular appliances from Xorcom and RedFone.

Digium vs. Xorcom

The combination of Xorcom Astribank (HW Channel Bank) and Xorcom TwinStar (Software for redundancy) is a similar application to the Digium R-Series. Below are the main differences between using the R-Series and Xorcom for redundancy:

Cost

• The Astribank hardware can run a customer up to four times the cost of R-Series appliances depending on the number of T1 interfaces that are populated in the Astribank. All eight T1 ports are included in the R-Series for $995 list price.
• The Xorcom TwinStar software is an additional cost. The open source tools used for the servers connected to the R-Series is downloadable and is free to use.

Hardware Platform

• The Xorcom Astribank channel bank has more parts, such as a power supply, that are prone to fail. Digium utilizes USB power from the servers to power the R-Series which is much less vulnerable.
• R-Series is 1U rack mountable and therefore takes up less space in the rack than the 2U Astribank.

Functionality

• The TwinStar/Astribank solution can only detect a hardware power failure if the Linux kernel has crashed. It cannot detect if Asterisk has stopped working but is with the R-Series solution (along with loss of power detection).
• If the R-Series were to lose power, the PSTN spans will continue to function through the same ports at the time of power failure. The Astribank will cease to operate at this point and communication with the PSTN is down.

Digium vs. RedFone

The RedFone gateways are another application where redundancy can be introduced to a customer location. By nature, gateways operate in a completely different manner than the R-Series for providing redundancy.

• A gateway is converting the PSTN TDM circuits to an IP format. RedFone uses TDMoE. Digium R-Series offers physical layer switching and does not convert the signal at all from one network to another.
• If the RedFone loses power, communications cease in their gateway.
• To deploy a redundant system, multiple gateways would need to be installed and therefore will be more expensive than the R-Series. Providing redundancy for up to eight T1’s will be at least four times the cost, just like the Xorcom solution, including several additional hardware pieces to manage in the datacenter.

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!

Of course, not all of us use the same types of computers, which can make things a little complicated for the SIP-uninitiated, to say the least. But using Session Initiation Protocol technology as part of a VoIP service will be worth the setup – especially considering just how easy SIP can be to set up.

Let’s focus on the world of Apple computer lovers today and recommend a few ways that you can set up SIP for Apple.

Apple Softphones

Using SIP allows for great communication services via the softphone, a simple technology that allows you to basically recreate a real phone on your computer. Because many of these phones are easy to download for free, the simple trick of it is to decide which phone is right for you. There are a number of different options available for Apple Softphones, including:

• 3CXPhone:  Free VoIP / SIP Softphone for iPhone.
• Switchvox Mobile:  Free; integrates Switchvox SMB phone system w/iPhone.
• Cisco Mobile:  Free Cisco Mobile 8.1 for iPhone
• Acrobits Softphone: SIP calls through Google Voice.
• Media5-fone: Free; compatible with leading enterprise IP-PBX and SIP servers such as Asterisk, FreeSwitch, and more.

Apple users will be happy with those downloads as they can be acquired over iTunes without any complicated instructions or setup requirements.

Of course, there’s more to SIP than just downloading the softphone. You have to understand the capabilities of SIP when you use a softphone, and you’ll find that there are more capabilities than simply dialing the correct numbers.

Because SIP technology over VoIP allows for so many computer-based possibilities, you’ll have a lot more options for communication when you switch your Apple over to SIP technology and start making your calls from the computer. Let’s explore these options.

SIP for Apple

The capabilities of SIP for Apple – and SIP in general – start with the basic two-way phone call. Anyone who downloads an Apple softphone will not miss out on that same experience. Simply dialing a number and having a voice chat is still possible – and in fact can be very convenient.

But there are other options to consider when an SIP session is started:

• Instant messaging can take place.
• Web page click-to-dial can take place.
• Voice-enhanced e-commerce can take place.

These options makes SIP essential in the telecom industry, of course, but can be just as useful if you’re going to be using SIP for a personal or professional use.

The easiest way to get started with SIP for Apple is to sign up to a VoIP service that offers SIP features – and then download a softphone like the ones listed above. You’ll quickly see how the new setup can change the way you do business and handle your daily life.

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!

When it comes to SIP softphones and open source SIPs (Session Initation Protocol), too many people are still left in the dark.

Luckily, this article is here to enlighten. We’ll provide a brief summary of SIP technology and what it might mean for both you and the way you handle your communications on a daily basis.

SIP Softphones

If you run your phone system using a VoIP service provider you’ll probably recognize just how easy it can be to ditch traditional phones in favor of softphones, which can be downloaded straight to your computer and often for free.

Being able to speak over a computer as opposed to a phone reduces everything from desk space being used to the money being spent on new phones! For this reason alone, this type of phone is often popular with start-up companies and new businesses. 

Using these softphones is just as easy and intuitive as using a regular phone – the calls and dialing are simply handled on the screen of your computer rather than on a piece of hardware like a telephone. While many people still prefer the actual feel and touch of an old-style phone, the capabilities of new softphones might ultimately be the way of the future.

Open Source SIP

With phone calls now capable of being handled through the Internet in the form of VoIP technology, SIP has become the go-to method for VoIP technology for signaling. Creating an SIP session allows for a range of services and possibilities including instant messaging sessions, IP centrix services, and web page click-to-dial.

Through Open Source SIP the possibilities of VoIP systems are becoming limitless and have long since eclipsed the capacities of many “hardphones.”

Traditional phones can still be of great use in an office, but many people like reducing their overhead (especially start-up companies) by using Open Source SIP rather than some of those traditional methods of communication.

Once streamlined, a lot of time and effort – and money – is ultimately saved.

Related Posts

• Hardphone or Softphone?
• Free SIP Softphone Roundup

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!

Open source Telephony continues to grow in popularity with businesses and institutions of all sizes. If your doing anything in the Telephony space, you need to be up to date with what’s developing in this space.

Highlights

Some highlights on the ClueCon schedule of presentations this year include:

• Philip R. Zimmermann, PGP creator, speaks on the VoIP Security Roundtable
• James Aimonetti, Sr. Engineer at 2600hz, presents, “Roll Your Own VoIP Cloud…In About 30 Minutes”
• Bill Sandiford, President/CTO of Telnet Communications,  presents, “IPv4 Apocalypse”

By Delevopers, For Developers

ClueCon focuses on the technical aspects of VoIP and telephony and is the best place to expose yourself to whats new and interesting in real world open source telephony applications.

Each day of the conference is filled with presentations and Q&A sessions with many of the leaders in the industry including hardware engineers, programmers and project leaders. 

Tickets to the event, at Chicago’s Sofitel Hotel, can still be had so register to attend ClueCon 2011 today.

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!

In short:  If you attend, you should learn a lot.

Quality VoIP Presentations With a Technical, Real World Focus

Share your experiences with brilliant open source applications developers and interact with the most influential open source telephony experts.  Enjoy three days full of technical VoIP and telephony presentations from expert speakers (including PGP creator Philip Zimmermann) that examine the bones of telecommunications.

The attention to detail is complimented with presentations about real world VoIP applications by developers and entrepreneurs using open source telephony software as a catalyst for business.

Intimate Atmosphere

Ever since the first ClueCon event in 2005, the same intimate atmosphere exists that developers and sponsors have come to appreciate. Expect to spend time with fellow professionals and listen to high-caliber presentations from carefully screened speakers.

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!

Zimmermann created PGP (Pretty Good Privacy) as an email encryption software package to ensure that your private matters stay private.  Distribution of his high strength software overseas violated US cryptographic export laws at the time resulting in a criminal investigation by the US government.

Charges were eventually dropped and cryptography export policies were relaxed.  Since then Zimmermann has been working on Zfone, a VoIP encryption protocol that aims to bring the same level of PGP’s email security to internet phone calls.

PC World named Zimmermann one of its Top 50 Tech Visionaries and is a continuous advocate for telecommunication privacy and security.  To hear his insights on the VoIP Security Roundtable, register for the ClueCon Open Source Telephony Developer Conference running from August 9-11, 2011 in downtown Chicago, IL at the Sofitel Hotel.

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!

In both cases, it’s like comparing green apples to red ones.  Both have their good points and both have not so good points.

With that said, instead of giving my opinion on which is better, Digium or Sangoma (Not Lohan or Sheen!); I decided to list the pros and cons of each card manufacturer and you can derive your opinion from that.

Read on for the the Pros and Cons.

Sangoma Pros

• Backplane technology allows you to add up to 6 a200 series cards or 4 a400 series cards in a server only using (1) PCI or PCIe bus.
• Half-height bracket support on all cards and ships with these brackets.
• Support for up to 8 T1 circuits.
• Sangoma Fax Sync solution
• Analog Sangoma A200 series ports glow red for FXO and green for FXS signifying to the user which ports are what functionality.

Sangoma Cons

• Not field upgradable in regards to hardware echo cancellation.
• Requires A200 cables for the A200 series cards to break out from proprietary RJ10 port to standard RJ11 jack.
• Sangoma A400 series cards use a proprietary DB-25 connection which you must use the A400 pigtail cable shipped with the card or purchase the Sangoma Y Cable to connect to RJ-21 Amphenol.

Digium pros

• 4 port (Digium AEX400) and 8 port (Digium AEX800) cards support standard RJ11 jacks, and 24 port cards (Digium AEX2400) support standard RJ-21 Amphenol.
• Field upgradeable echo cancellation modules (can be added after the purchase).
• Created by and supported by the makers of Asterisk™

Digium Cons

• Digium Analog series cards do not support half height brackets.
• Expandability may require the use of additional PCI or PCIe buses on the server motherboard (No backplane technology).
• T1 cards only up to 4 T1.
• Pins on cards used to connect FXO/FXS and echo cancellation modules have a tendency to get bent very easily and break.

Still Undecided?

 Now that you have the pros and cons, I hope this helps you make an informative decision when deciding to pick your card.  If you are still unsure, always feel free to contact me directly and I will be glad to help.

Ramon Perez

rperez@voipsupply.com

Toll Free Direct (866) 675-8455

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!

We all know the capabilities and endless feature set of Asterisk. It’s a powerful, software based PBX, that’s possibilities are endless. Asterisk is distributed in a number of ways, the first being the open sourced distribution that is command line driven. This distribution, 1.6.2.11 being the latest stable version can be downloaded from here for free, then compiled with a version of Linux, such as UBUNTU or Centos, and then administered from the command Line CLI interface.

The Flavors of Asterisk

Then we have your open sourced asterisk-based distributions freely available to the public for download. These versions would include trixbox CE, Elastix, FreePBX, PBXinaFlash, and Asterisk Now, which just names a few. The separator between these software applications as compared to the command line asterisk distribution is that while asterisk source code is running in the background, the majority of configuration changes can be made via a Graphical Web GUI interface from both a user and administrators perspective. These software applications also allow for asterisk cmd line access or CLI access in order to make changes if required. Companies such as these have taken the asterisk source code and “rounded” it to their own unique application.

And last but not least, we have our commercial asterisk based PBX’s. Most notability would be Switchvox SMB or SOHO software which is asterisk based, but no access is granted to the asterisk CLI. Rather all configuration changes are done via a web GUI interface. Software from Switchvox is considered a “paid” commercial product, which distinguishes itself from any of the open source based distributions mentioned above.

With all flavors of asterisk, each meets its own unique need and fulfillment in individual VoIP deployments. When it comes to purchasing an IP PBX, you have your choice of asterisk software and also hardware to go along with it. In all regards, whether you choose the free open source cmd line version of asterisk or pay for Switchvox’s award winning SMB software, all solutions present you with a powerhouse feature set and customizability to meet your unique business needs. A short list if asterisk’s feature sets can be found here . The list is too long to fit in this blog.

Linux is a Dominant Force

Until most recently, the majority of IP PBXs being deployed were of an asterisk flavor based on a Linux operating system. While companies such as trixbox, Elastix, Switchvox, and many others have made great efforts to take the Linux programming piece out of the picture, the solutions are still Linux based meaning when it came time for your choice of hardware or server to run your asterisk solution, you needed to make sure the motherboard and peripheral components were compatible. See my previous blog posts relating to Open Source Hardware Requirements for more information. If you were not designing your own server platform, of course you could always purchase a Switchvox appliance for your SOHO or SMB solution, or a Rhino or Phoneboch’s appliance preloaded with your desired version of trixbox CE, Elastix, or other open sourced based software. This however adds to the cost of your solution.

For the following comparison, I would like to focus on Switchvox SMB software and its comparisons against a new windows-based PBX, 3CX. The reason for this is that both solutions are “paid” commercial software based applications. Pitting these 2 up against one another would make the most sense from a cost, feature set, and positioning standpoint, so onward to the comparisons.

3CX Compared to Asterisk

3CX is a Windows based IP PBX platform that is becoming very popular in the VoIP world. A few benefits of 3CX include, “its windows based”, most of us are used to Windows based operating systems and applications; it is completely software based and allows you to use existing Windows servers in your existing environment if available. For a complete list of compatible windows based operating systems, see here. Below are a few more benefits, 3CX has to offer:

-Windows Application Integration support- applications such as Microsoft Exchange, Outlook, Unified Messaging , Microsoft CRM , and SalesForce CRM are all fully supported on 3CX version 9.

-Feature rich- functions and feature sets of 3CX version 9 are very comparable to an Asterisk distribution. See here for a complete set of features. Please look at the “paid commercial” versions.

-Free and Paid commercial versions- are available to experiment with and test. Free versions of 3CX software are easily upgradable to any commercial paid version located here, through a unique activation license key provided to you via email after your purchase. You can then apply it directly from the 3CX admin GUI configuration. Updates are instantaneous after you apply them. They are based upon the maximum number of concurrent calls you will have on your solution, and can be easily upgraded to higher amount of simultaneous calls if needed. Once again, changes are done via the web GUI and are instantaneous.

-User Dashboards– the 3CX assistant provides each 3CX user a PC based presence application where they can easily set their calls rules, answer calls; listen to voicemails, and chat. More information on 3CX Assistant can be found here.

-Hotel Billing and PMS integration- On top of their paid software platforms, administrators can add the 3CX Hotel Billing, or the full-blown Hotel PMS integration module to easily integrate Hotel PMS platforms with 3CX. This is a huge hit in the hospitality industry. More information on this support can be found here. For pricing on the software add-ons, see the billing and PMS products on VoIPSupply.com.

-Non proprietary, SIP Standards based- Since the 3CX solution is based upon SIP protocol, administrators have the ability to use any SIP based endpoints such as IP Phones, Gateways, ATA’s ,and DECT Phones . Furthermore 3CX has fully tested a large number of these endpoints and offers full support including auto-provisioning. See here. If you are looking to add some PCI cards to the solution, you will want to go with a Sangoma PCI FXO or T1 Netborder Express card with echo cancellation.

-Smartphone Apps- 3CX offers and fully supports a soft-phone application specifically designed for Android’s, and is beta testing a new app for Apple products such as the iphone, ipod, and ipad running IOS 4 or higher.

Now onto the Switchvox SMB platform.

-Third Party Application Integration support– Switchvox SMB integrates very easily with Sales Force CRM, Sugar CRM, Fire Dialer for Firefox users, Notifier Suite to integrate Switchvox to outlook and other Windows applications, and finally an Outlook plug-in to allow click to call from your outlook contacts. More information can be found here. If that’s not enough, Switchvox offers a developer API

-Feature Support- I always like to refer the SMB platform as Switchvox’s “Big Daddy” PBX, the reasonin behind it is simple; it is jammed packed with features for both users and administrators. A full list of SMB feature set can be found here. Make sure your looking in the SMB column.

-Multiple software versions available- If SMB is too much for you, you can opt for Switchvox SOHO or HOME versions which are more limited in functionality and cheaper in costs as compared to the SMB solution. Also, Digium offers a FREE 60 Day Trial of the full-blown SMB software, so you can “test drive” the actual SMB solution before making your purchase. If you are interested in test driving the SMB solution, contact our dedicated Sales Representative Arthur Miller at 716-250-3871 to discuss the details.

-User Dashboards- Personally, The Switchvox Switchboard, ONLY available with SMB is one of the most intuitive presence applications on the market, and it all comes bundled with your SMB solution. From the Switchboard, users can drag and drop calls to other users, see other users real-time call state, access VM messages, customize to see Google Maps, integration with CRM accounts, Queue status, CDR, Chat, and the list goes on. If you would like to see the switchboard in action, just watch it here.

-Non proprietary SIP based endpoints- While SMB has built-in auto provisioning support via is Provisioning Token system for all Polycom and Snom IP Phones, it is SIP based, so any SIP endpoint including IP Phone, gateway, ATA, etc will function. If you are looking to add some PCI cards to the solution, you will want to go with any Digium PCI card for your FXO/FXS, or T1 connectivity. Echo cancellation models should also be considered.

-Smartphone Apps- Digium Switchvox offers a Smartphone application for both apple based products and Blackberry’s. For you Droid users, sit tight, you will get yours soon. The application allows users to create and modify their call rules, access VM, and call into the Switchvox PBX. More information can be found here.

So what do you think? I know this is just a short comparison between these two systems, the list could go on and on, and hey maybe one day, I’ll write a book about PBX comparisons. So cost aside, which one would you choose and why?

This information was originally posted on the VoIP Insider blog.

Need VoIP Phones or a VoIP Phone System? Checkout VoIPSupply.com!