The Hidden Costs of an IP PBX System

Moving to an IP PBX system can be financially advantageous, resulting in lower long distance fees, lower hardware costs, and access to higher-end features for very little extra money. The salespeople who offer the systems are quick to point out these advantages, and they are correct—but before you sign on the dotted line, make sure you have the whole picture. There are other costs involved besides the retail price of the equipment, which may not be immediately obvious. Even with all indirect or “hidden” costs factored in, IP PBX is still often a wise decision, but it pays to be informed, and be aware of the big picture.

Bandwidth

One of the first hidden costs you could potentially face is the cost of additional bandwidth. If you are not already using VoIP, you must realize that once your IP PBX is installed, your voice traffic will be moving over your Internet connection, as opposed to the phone company’s copper wire. This will place an extra burden on your existing Internet connection, and you may need to upgrade for additional bandwidth. If you have a small office with only a few phones, and you have a DSL connection to the Internet, you may be able to get away with keeping your current bandwidth, but an upgrade may be in order, depending on how many phone lines you have and what your simultaneous usage may be. You may need to upgrade from DSL to a fractional T1, or if you have a fractional T1, you may need a full T1, and this will represent additional monthly cost.

New phones

If you’re moving to VoIP, you’ll need IP phones, since your old phones won’t work and won’t support the advanced features you’ll get from your IP PBX system. In many cases, this is a hard decision, since a large office may have tens of thousands of dollars tied up in telephones which are perfectly usable. You have a choice between purchasing all-new IP phones, or purchasing adapters for your existing phones. The adapter option may save some money, but it may be defeating the purpose of deploying the IP PBX in the first place. The modern IP phones are built with IP telephony in mind, and are ready to take advantage of the many new features that would not be supported on your old phones with just an adapter.

Installation

If you’re running an on-premises IP PBX, you need to either have skilled staff on site, or use the services of a professional who will install the IP PBX at your facility. Ideally, this installer will be able to switch over to the new system in a fairly transparent manner, so you are not faced with any substantial downtime; and in addition, they should also be able to take time to train you and your staff on how to use your new system.

Integration

The high-end features of your IP PBX system will allow you to integrate your phone system with many existing applications. For example, your phone system could be integrated with your customer database, so that your call agents get a “screen pop” whenever a call comes in, showing not only who is calling, but their entire order history and sales notes. This type of integration adds to productivity, but may require some custom integration by professionals who understand both the IP PBX system, and the other applications with which it must work.

AccessDirect Virtual IP PBX

AccessDirect Virtual IP PBX system comes with all the features that are usually found only on high-end PBX system owned by large companies. IP Virtual PBX lets small and medium businesses to establish their business presence anywhere in the United States. That means you can operate your business from your home or from any isolated region without relocating to the place where you are interested to establish your business presence. Consequently, our Virtual IP PBX eliminates the operating expenses associated with traditional branch offices, and unaffordable long distant call rates.

VoLANs IP PBX

VoLANs VL series IP PBXs are all-in-one solution especially designed to meet the various needs of SMEs and to enable VoIP service providers to quickly and cost-effectively launch basic business VoIP services.
VL series provides an effective and complete communications solution for cost-conscious companies. It improves productivity and enhances customer care, while reducing capital expenses and operating costs.
VL series is a communication system that integrates two essential business operations into one simple system: a feature-rich telephony system and a robust data network system.
The system provides enterprise-class features, such as custom call routing, various call services, remote user access and site-to-site connectivity. It also supports remote users and you can connect multiple VL series IP PBXs to create one telephony system across multiple sites with VPN mechanism.

Wangate PBX

Wangate PBX systems allow to connect both digital (SIP/IAX2), analog (FXS/FXO) or digital (ISDN BRI/PRI and GSM) with a single product.
Our PBX can be expanded with internal gateways for analog/digital interfaces resulting in lower costs and better voice quality, as they're engineered by us. You can also use standard external gateways (SIP/IAX2) for FXS/FXO/GSM/BRI/PRI (etc) connections.
Based on the open source asterisk platform and open source software, our PBX do not impose fixed limits on the number of extensions, gateways, trunks etc.

Standard Features

All Wangate PBX use the same firmware and are compatible with one another. Even the smallest PBX1 is a full blown PBX with no limits and advanced features:

• Ring groups
• Voicemail for every extension with email capabilities.
• Music on Hold (MOH)
• Automated Attendant (IVR menus)
• DISA
• Active channels monitoring in real time
• Multiple dial plans and access control lists
• Conference rooms

Technical Specifications

Wangate PBX differ in their hardware interfaces but they all share the same basic specifications:

• SIP/IAX2/RTCP protocols supported
• Unlimited VoIP trunks. 
• Unlimited VoIP extensions.
• Echo cancel with hardware DSP
• Jitter control
• Full list of VoIP Codecs (G.729AB, G.711, G.723, G726, iLBC, Speex).
• CNG/VAD
• DTMF Tones
• Support for FAX/Modem (analog/isdn lines) and T.38 pass-trough on VoIP lines.

onecomms PBX

The onecomms PBX is a business IP phone system which includes many features as standard
The PBX is available in a 1U rackmount, a 4U rackmount, a mini wallmount, or HP Server based floor-standing chassis for convenient installation into any business environment.

Based on standard SIP architecture, the PBX is compatible with most VoIP applications and hardware.
With the addition of a line-card, the system can also be connected to existing landlines, as well as VoIP lines. This gives multiple incoming and outgoing calling routes, as well as backup for any line which may fail.
Running on standard RJ45 Ethernet, the PBX will integrate into most companies network infrastructure without the need for additional cabling on the premises.

Some of the many features include -

• Free calls to laptop users on the road or at home
• Free site-to-site calls
• Call Recording & Monitoring
• Unlimited Extensions
• Unlimited Lines
• Voicemail and voicemail-to-email forwarding
• Call Queuing
• Auto Attendant/Call menus
• Call ringing groups,
• Call diverts
• Least Cost Routing
• Automatic switchover for night time & weekends
• Customisable Music on Hold
• Call Logging & Reporting
• CTI (on-screen pop-ups)
• CRM Software
• Links into Outlook, ACT! or any other database software
• Works with any Standard IP SIP phone
• Simple/Plug-and-Play Install
• Low Cost installation
• Enables remote users
• Paperless Faxing
• Fax-to-email
• Support for Videophones

PortaSwitch

PortaSwitch is a software based telecommunication services and subscriber management platform that allows VoIP service providers, carriers, ISPs, and modern communication network operators to unify voice, data, Internet and fax traffic within a single
converged network.
PortaSwitch is built around a comprehensive converged VoIP billing software platform and includes a class 4-5 SIP softswitch and media application servers. PortaSwitch provides access to the complete hosted IP PBX or IP Centrex functionality and the ultimate feature set as a calling cards or VoIP wholesale platform.

Voicetronix OpenPBX

Voicetronix's OpenPBX is Open Source and is entirely written in Perl. It runs on Linux and interacts with a range of multi-port analogue FXO, FXS and digital (Primary Rate) line interface PCI cards via CT Server, a Perl based client/server library supporting Voicetronix hardware.

OpenPBX a full function, web enabled PBX application, suitable for small office installations and can scale to large call centres. Features include a web based user and management GUI, unlimited Voicemail, Hierarchical Auto-Attendant, Automatic Call Distribution ACD, Least Call Routing (LCR), Music on Hold (MOH), Call Display Records (CDR), unlimited huntgroups, call transfer, call parking, call baring. It has the ability to offer 3 way call conferencing and by leveraging the power of the desktop it offers voice to email, click to dial and transfer of calls.

Brekeke PBX IP Phone System

Brekeke PBX is an award-winning Business IP Phone System that provides robust, high performance, and intelligent IP-PBX (PABX) functionality.

Supporting the industry standard--Session Initiation Protocol (SIP)-- Brekeke PBX delivers a highly compatible product for migrating existing applications and services. Brekeke PBX comes in two formats: Single-Tenant (Basic & Pro Ed.) for premises-based IP-PBX installation, and Multi-Tenant for Hosted PBX platform. The Brekeke PBX is highly scalable and supports up to 2,000 users and more. Brekeke PBX supports Microsoft Windows and Red Hat Linux.

AltiGen IP PBX

AltiGen Communications VoIP Phone SolutionsSRMS Network Technologies is a full-service managed solutions provider and AltiGen distributor.  SRMS will prepare, recommend, install, train, support, manage, and maintain AltiGen PBX Phone Systems. Using a single-server solution configuration and installation of your phone system is done quickly and efficiently.AltiGen’s IP PBX Phone System is the world’s most widely deployed 100% Microsoft-based Unified Communications system.    Unlike the network switch-based VoIP systems, AltiGen’s open software-based approach offers maximum functionality, flexibility, and scalability – all without requiring expensive, proprietary hardware systems.

VoIPCortex IP-PBX

ipcortex is set to radically change the PBX landscape with the second major release of its VoIPCortex IP-PBX platforms. Enhancing the capabilities of its mature and widely deployed platform, this release builds upon the functionality of the existing Pro and VoIPCortex rack mount units and adds an entirely new Compact VoIPCortex unit. Optimised for SMEs, the Compact PBX allows smaller organisations to discover the benefits of VoIPCortex IP-PBX solutions.
VoIPCortex units are flexible, open, feature-rich, and above all simple to configure, leading to much reduced deployment costs for installations of all sizes.
All VoIPCortex systems now include advanced features such as voicemail with unified messaging, call records with billing data, IVR menus, conference bridge, hunt groups, hot-desking capabilities, music-on-hold, call queues and soft-fax - all configured through an intuitive web based UI. Auto configuration of Aastra, Elmeg, Linksys, Polycom and snom ip phones is a key feature for speedy system installation.

AskoziaPBX Open Source PBX System

AskoziaPBX is a complete PBX (Private Branch Exchange) on a small live CD that can be installed to your hard drive. It makes Asterisk, the number one open source PBX, install in minutes. Coupled with a little bit of telephony hardware or a SIP phone and a VOIP gateway account, you can have a complete phone system running in less than half an hour using AskoziaPBX.

IP PBX

For PBXs, IP telephony has crossed over from an emerging technology into the mainstream choice for business voice. The initial technical and operational challenges faced by IP PBXs are behind them, ROIs are proving the economics to be solid, and the productivity gains associated with the technology are and will continue to exceed those that could be realized with traditional telephony. User acceptance of IP telephony is high, with the sales of IP lines in the first half of 2007 solidly continuing to outpace those of traditional lines, as the move to IP telephony continues.

VoiceAxis PBX

VoiceAxis is multi-tenant PBX and VoIP management software designed to enable Service Providers and Enterprises to install and manage a large scale Asterisk implementation. The software controls a unique VoiceAxis Asterisk Cluster environment which utilizes Asterisk as the feature server or “SoftSwitch”. VoiceAxis works with both Asterisk Business Edition and the Asterisk Community Edition (asterisk.org), and leverages several other open source and commercially supported technologies.

PBX in a Flash

PBX in a Flash is the Lean, Mean Asterisk Machine designed to meet the needs of hobbyists as well as business users and VARs. You'll have a high-performance turnkey Asterisk PBX that's easy to upgrade with dozens of add on scripts to provide virtually any feature you can imagine. With PIAF you can choose from tons of Nerd Vittles and FreePBX applications that install in under 15 seconds: AsteriDex, Weather Reports, News Feeds, Email by Phone, Telephone Reminders, and many more. You add features when you need additional functionality. Otherwise, Just Say No to Bloatware!

Virtual PBX Provides Winning Service for New York International Spirits Competition

Posted on October 26, 2010 by C. Sensendorf

When you are hosting the world’s foremost spirits and wine competition, it is crucial to have a phone system that can back you up all the way. Luckily, Adam Levy, the host and founder of the New York International Spirits Competition (NYISC), was well prepared to recognize and award the best spirits and wines in the world.

Levy easily and efficiently ran the NYISC with the Virtual PBX® business-class phone system. Already familiar with the best, he knew VirtualPBX would be the perfect system to keep him in contact with a multitude of judges and vintners, and manage numerous calls from any location. In addition, receiving all the award submissions was no problem with the built-in faxing capabilities offered by VirtualPBX.

With Virtual PBX, Levy was able to significantly reduce overall costs and have access to vital features including follow-me calling and inbound fax. He estimates the new system saves the company around $980 a year on phone charges.

“After using Virtual PBX at a previous company, I knew that they offered a great service that would be easy for us to implement,” said Levy. “Today, our team relies on Virtual PBX to retrieve all of our voicemails and faxes, even when we are in completely different locations. In particular, we value the ease-of-use and varied feature set, which allows us to work out of our home offices and on-the-go when we’re at tasting events.”

The industry’s leading spirit experts are a huge part of NYISC’s overwhelming success. To compare and select the winning spirits, they had to be in constant contact with buyers, sommeliers, restaurant owners, distributers and importers. They never missed a call or fax as the virtual PBX system effortlessly routed the hundreds of entry forms as well as numerous calls to the appropriate line.

“A majority of our customers are on-the-go and require flexible resources to help them keep up with the needs of their clients, regardless of their location,” said Paul Hammond, president and CEO of Virtual PBX. “With the industry’s strongest set of features included in our standard plans, we give customers like the New York International Spirits Competition a cost-effective and powerful business phone system that delivers peace of mind.”

Our congratulations to Adam Levy and the New York International Spirits Competition. We’re proud to be involved with such a prestigious event!

Filed under: follow-me calling, general telephony, phone system

Virtual PBX Adds VoIP Phones and New Flat-Rate Plans!

  Virtual PBX Adds Hardware VoIP Phone Support

The time has come — Virtual PBX, provider of the no-compromise business phone system, announced support for hardware VoIP Phones along with new flat-rate plans, fortifying their lineup of industry-leading plans and services. With this, Virtual PBX now controls the phone call from the inbound phone number ALL the way down to the phone on the desktop. You can now have a truly complete phone system integrating phone numbers, free long distance and IP phones into one cost-effective solution.

We now offer two award winning plans from which you, the customer, can choose: the new flat-rate plans and the original usage-based plans. Even better, you don’t have to be on the flat-rate plans to use the VoIP phones – we’ve set up the system to support the phones on either plan. So feel free to pick the plan that works best for you and your business, and also decide how you would like the system to reach you. You can be contacted through the VoIP phone(s) and/or contact phone numbers via a landline or cell phone. The freedom to choose is in your hands.

We also offer you the freedom to save! With Virtual PBX you can simplify your life and your budget by cutting the ties with your landline carrier. There is really no need for a landline anymore seeing as how you can make and receive phone calls using a VoIP phone and your Virtual PBX service. So go on ahead and save that money each month with Virtual PBX Complete. With our phone service you can enjoy a savings of anywhere from 40-80% in traditional phone system and landline costs – that is a huge savings! Given that telecom costs are a major expense in everyday business, Virtual PBX provides business communication that is affordable, easy and accessible from wherever you need to be with a low monthly fee and no contracts or setup costs!

Please see the plan details below and save money with the plan suited for you:

Flat-Rate Plans:

These are your traditional “unlimited” plans that so many companies try to market. However, Virtual PBX does it right as there is a fair-use policy in place (much like every other VoIP phone service out there) that allows 5,000 minutes of inbound minutes per extension, aggregated at the account level. So a 5-user system would include upwards of 25,000 inbound local minutes. That’s amazing. Most companies won’t touch that. Yet, they can have this system for around $150 per month, and that already includes the e911 fees, Federal and FCC taxes. The flat-rate plans are as follows:

1-User: $39.99/mo4-User: $23.99/mo/ext10-User: $21.99/mo/ext20-User: $19.99/mo/ext50-User: $19.99/mo/extExtra Extensions: $24.99/mo

Usage-Based Plans:

These plans are what Virtual PBX used to pioneer the hosted phone service market. On these original plans you only pay for the minutes used each month. As the plans go up, you receive a larger block of free minutes and the price for additional minutes becomes cheaper. These plans come with unlimited users and all the same features as the flat-rate plans. Virtual PBX has the following usage-based plans:

VPBX-5: $9.99/mo (300 minutes, extra minutes are 6.5 cents/min)VPBX-10: $24.99/mo (600 minutes, extra minutes are 5.8 cents/min)VPBX-20: $44.99/mo (1,000 minutes, extra minutes are 4.7 cents/min)VPBX-50: $94.99/mo (2,500 minutes, extra minutes are 4.4 cents/min

VoIP Phones

Virtual PBX now supports hardware VoIP phones connected to the service. This is a superb addition as now you can make phone calls and show your Virtual PBX caller ID on every outbound call. And you don’t have to use a cellphone or landline to make that outbound call. All you need is an Internet connection. Virtual PBX sells the following phones that arrive pre-configured and ready for use:

Polycom SoundPoint IP 321: $99.99 (single ethernet port)Polycom SoundPoint IP 335: $139.99 (ethernet pass-through port; backlit display)Polycom SoundStation IP 5000: $599.99 (conference phone)

With Virtual PBX there is no need to get phones from one provider, calling plans from another, and features from somewhere else. Get it all in one place. If you are looking for a complete, no-compromise business VoIP solution for your business, Virtual PBX is the answer.

Filed under: ACD queue, auto-attendant, call recording, Call tracking, Call transfers, Call whisper, CallerID, conferencing, follow-me calling, hunt group, mail only extension, music on hold, phone system, virtual receptionist, VoIP Tagged: | caller id, extensions, features, hosted pbx, hosted phone system, IP PBX, IP phones, pbx, phone system, phones, Polycom, SIP VoIP, small business, small business phone system, virtual pbx, Virtual PBX Complete, virtual phone system, virtualpbx, VoIP phones

Hosted VoIP Phone Systems vs. TDM: Who Really Wins on Cost?

The cost savings of a hosted VoIP phone system often comes under scrutiny from some of the remaining providers of TDM, or “traditional”, phone systems. You know the kind, those broom closet phone boxes of yore. Through the use of a narrow lens, they push out misinformation that may lead small business owners to think that a TDM solution will somehow save them money over VoIP.

This is, however, an act of desperation on their part – not a reflection of reality. By choosing the scope of their argument, they can stack Apples v. Oranges and shine a deceptively attractive light on their systems. Zoom the camera out a bit, get the whole picture, and you’ll see why smart businesses have been abandoning TDM solutions in droves for the advantages of hosted VoIP.

The Classic TDM Cost Savings Argument

Let’s start with the argument for a TDM phone system. In a nutshell, these providers cite the benefit of their systems as being the fixed cost to own, e.g. pay for the PBX equipment and the installation, and it’s yours to own for life. They then compare that high fixed cost to a month-to-month hosted VoIP service like FreedomIQ over some extended period, say, 24 months, to say the TDM phone system carries a smaller price tag in the long run. Conversation over, right?

Not even close.

Where the TDM Argument Falls Apart

Hosted VoIP is a managed service, and much of the cost savings is abstract from the direct cost of equipment and ongoing service. Any savvy business owner knows that the price tag they should care about, the real price tag, is one that includes both direct and indirect costs to get the Total Cost of Ownership (TCO). With that in mind, here’s what you’re not seeing with a direct business VoIP vs. TDM cost-of-service comparison:

1.  Deprecation of Hardware

TDM PBX equipment, like a car, loses value the minute you deploy it. The notion that you’re going to “own” your PBX isn’t exactly exciting if it doesn’t hold any value 6 – 12 months down the road and you’re back in the market. Worse still, if it breaks down while you “own” it, that means it’s your responsibility to pay to have it fixed or to buy a replacement. Even if you’re covered by a hardware warranty you’ve purchased, that cost needs to be a part of their big picture comparison.

With a hosted VoIP phone system, you’re not dropping capital into hardware that’s going to become obsolete. There’s no buying replacement parts for your wall switch and no risk that a year from now you’re going to be stuck with a 5-figure paperweight. Not to mention the fact that the equipment delivering your hosted service from remote is enterprise-grade and fully-redundant, something you’re not going to get from a comparable TDM solution.

2.  Scaling Up Your Business

TDM phone systems are notoriously inflexible to the growing needs of your business. With this kind of phone solution, you’re likely to fall into one of two camps: the over-paying, or the under-provisioning. That is, you either buy a PBX that is too big for you right now, thinking that you’ll eventually grow into it, or you buy a PBX that meets your short-term needs, understanding that you’ll have to figure something else out should your needs change. Result? You’re paying more than advertised, either right now for the privilege of future expansion, or down the road when the necessity to upgrade your PBX hits you like a ton of bricks.

Hosted VoIP phone systems are scalability machines. By taking your hardware off-premise and having phones interconnect and transmit calls via the Internet, you’re removing the limitation on what size of PBX you can have. You won’t need to overpay for more phones and capacity than you need or buy a brand new PBX within a year because your staff size has grown by 20%. With hosted VoIP, you can add phones to your system at any time and from anywhere with no extra costs to scale.

3.  Paying IT Staff / Maintenance

The coup de grâce. Owning your own TDM PBX again brings the responsibility of management, maintenance, and configuration in-house. That means some percentage of your IT staff budget is getting allocated to making sure your phones are always up-to-date and in working order. Even if you’re talking a tiny fraction (5%) of just one IT employee’s time, you’re talking about thousands of dollars each year to manage your own PBX.

Part of what you’re paying for with a hosted VoIP PBX is, essentially, the outsourcing of this responsibility. We maintain the equipment that provides you with service. We keep your phones updated to the latest firmware. We help you configure your service and troubleshoot your setup with free U.S. technical support. Meanwhile, your IT staff gets to focus on issues more critical to advancing your business, and the money you save (or make) by keeping them on task all but pays for your hosted VoIP phone system.

Be sure to do plenty of research and look at the big picture when considering what type of phone system is right for your business. Look beyond the price tag to what a system provides in flexibility and time saved, as those ultimately lead to keeping dollars in your pocket.

Hosted VoIP Lite: Getting Your Feet Wet

Making the move to a new business phone system can be downright scary. The communications technology that a company chooses has become so critical to business efficiency, that one IT misstep can easily set you back weeks in terms of productivity.

This is just one of the many reasons we’ve seen businesses turn to a hosted VoIP phone system. With hosted VoIP, there’s no big upfront commitment. No massive equipment or servers to finance, no infrastructure changes, no heavy wiring. You are free to try out the system with as small of an initial deployment as you’re comfortable taking on – even just a single phone – before deciding if it’s right for your business.

The Way It Used to Be (Traditional PBX)

Before the days of hosted VoIP, your IT decision maker’s best friend was research. They’d spend hours reading white papers, watching presentations, or attending webinars on why they should choose one PBX vendor over another. The prospect of trying out a phone system live on your premise with little to no commitment was laughable. There was simply too much that needed to be done to get it to work, and no vendors that would do that work without expecting to be paid in full.

It’s not the vendor’s fault, really, but a problem in on-premise technology itself. All premise-based PBX systems require equipment and wiring that takes a solid investment of time and money. If things don’t work out, whether the technology simply isn’t a good fit, the service isn’t reliable, or the customer support is lacking — it’s too late. You’ve already signed on the dotted line and there’s no going back.

But over the past few years, hosted VoIP has changed the game. Speculative research and sales fluff has given way to live on-site demonstrations, demo phones, and the advent of the phased-in VoIP phone system.

The Phased-In VoIP Phone System

As businesses become more savvy about VoIP phone system technology, we’re seeing more and more opting to get started with a small set of VoIP phones (or even just one). These phones use your business’s existing data connection, requiring no invasive wiring, no premise hardware, and can be up and running in the course of a single visit.

This way, your key players like the head of IT, the CEO, or even a small department can serve as guinea pigs without impacting the day-to-day operation of the entire business. These limited roll-outs can often be fully integrated with your existing business phone system for a seamless phasing-in of phones over time.

And it doesn’t take long. As you experience important factors such as call quality, reliability, potency of features, and ease of use, the decision largely makes itself. If you’re not happy with the technology, it’s easy enough to pull the plug. If you’re confident in the technology, the ability to add new phones anywhere at any time helps you ramp up your deployment to include additional departments and even additional locations.

Are you ready to try hosted VoIP for yourself?  Give our consultants a call at 888-955-3520 to have a risk-free, no-obligation conversation about how you can test out FreedomIQ for yourself with a small deployment.

How to Convince Your Boss to Try Hosted VoIP

IT managers working for an SMB have a tough gig: maximizing the return on every IT dollar spent. With tech budgets shrinking across the nation, there’s an urgent need for every IT product or service a company chooses to deliver higher value at a lower price. Fortunately, this need has resulted in the emergence of a host of cost-effective, productivity-enhancing technologies. Among them, the hosted VoIP phone system leads the charge for smart business communications.

But, if you’re an IT manager, you already knew that. You’ve done your homework. Now it’s time to convince your boss, who may be coming from a different perspective and have different priorities. Here are some tips on how to present a hosted VoIP solution that will get the C-level to sign on the line that is dotted.

1. Easy on Cash Flow

Why not lead with the knock-out punch? Hold the pretty explanations and come out of the gate with the fact that the upfront capital expense of a hosted VoIP PBX is tiny compared to any other professional phone system. For equipment, you’re paying for the cost of your VoIP phones and maybe a couple VoIP routers. There’s no $10k+ box-in-the-wall or clunky on-site server to worry about financing. Hosted VoIP is the professional phone system your boss can afford even if he or she can’t afford a professional phone system.

2. Conserve Your IT Staff Resources

With a managed service like FreedomIQ, you don’t need to squander your precious time wondering what the blinking light on your PBX means. You don’t have to thumb through manuals or search for tech guides online should a phone go on the fritz. Your boss can keep you and your IT staff on business-critical development full-time because you have a hosted VoIP service provider to take care of these things (or prevent them from happening in the first place).

3. No Investment Risk

Remember when we were talking about the minimal investment required (#1) to get hosted VoIP service? Well, not buying that wall switch or server in the first place means you and your boss don’t have to worry about what to do with it a year or two down the road. You know, the time just after its hardware warranty has run out and it’s either a) quickly turning into a steaming pile of junk or b) you’ve outgrown it to the point where it’s practically tinkertoys relative to the needs of your business. Hosted VoIP’s equipment-free nature provides an ease of scalability that saves your boss from the embarrassment of a 4- or 5-figure PBX paperweight.

4. Strong Productivity ROI

Saving money is great, but what about your employees? Won’t somebody think of the employees? Enhanced productivity is where you’ll find that hosted VoIP is no slouch. Even though you’re paying a fraction of the price, you’re getting all of the advanced features and functionality of an enterprise phone system.

You get unlimited local and long distance calling, a professional auto attendant, call forwarding with ACD routing, voicemail service, Internet faxing, web-based settings and call reports – the list goes on. But the real productivity kicker is when you show your boss how FreedomIQ integrates into the programs his or her employees already using, like Outlook and Salesforce CRM. With this feature, FreedomIQ works seamlessly with your existing business environment, helping employees be as efficient as possible.

Virtual PBX Honored to Receive its Seventh Award for Open VoIP Peering

Virtual PBX, the pioneers and market leader of the virtual phone system, was honored to received their seventh award from Technology Marketing Corporation (TMC) for their innovative Open VoIP Peering service. This particular award was presented by TMC’s NGN Magazine, the leading resource for communications service providers.

The NGN Leadership Award recognizes the growth and evolution of applications and services designed for next-generation networks and the enabling technologies that make them possible. Virtual PBX was recognized in the Services category.

“This recognition of Virtual PBX’s Open VoIP Peering service is a testament to the company’s continued commitment to providing customers with innovative solutions that meet their demanding business needs,” said Paul Hammond, CEO of Virtual PBX. “Being selected as an NGN Leadership Award winner is truly a mark of distinction and further validates Virtual PBX’s leadership in the VoIP market.”

So far Virtual PBX has received 3 product of the year awards, 2 leadership awards, 1 Product Innovation award and 1 telephony excellence award for the Open VoIP Peering solution they released in November of 2008. No stranger to winning awards, Virtual PBX’s 7th win comes on the heels of their impressive ‘2010 Best Telecommunications Services in San Jose’ award.

The many accolades that Open VoIP Peering has received truly are a testament not only to the company’s strength and dedication in bringing to market the most innovative product conceivable, but also the products immense popularity with Virtual PBX Customers everywhere.

The Open VoIP Peering Solution stands out from the masses, being the only VoIP solution to allow users to choose which option they like on a call-by-call basis to meet their individual needs. In addition, unlike competitors who charge extra for proprietary digital-enabled phone extensions, with Virtual PBX there is no additional cost for accessing soft phones.

“Virtual PBX was recognized for its exceptional work in advancing NGN services and technologies. Open VoIP Peering has proven exceptional and its innovation has contributed to the transformation of the industry,” stated Rich Tehrani, CEO, TMC. “Congratulations to the entire team at Virtual PBX, and I look forward to more innovations in the coming year.”

The 2010 NGN Leadership Award winners can be found in the May/June 2010 issue of NGN Magazine.

Virtual PBX Adds Call Recording – For Free!

We are excited to announce every Virtual PBX system now includes call recording in all standard plans at no added cost for customers.

“This is the latest in a series of enhancements that Virtual PBX has rolled out this year,” said Greg Brashier, COO of Virtual PBX. “We’ve had requests for call recording from our client base and felt it was something a majority of our SMB clients could use. Since we innovated the entire hosted PBX space, we feel it is necessary to lead by example and continue to enhance the service.”

With the newest Virtual PBX call recording feature, any or all calls can be recorded, saved, and played back later. There are infinite ways companies of all sizes can utilize this tool to enhance their business such as capturing calls for later review, assisting in employee training, monitoring customer service agents, as well as meeting legal requirements.  Greg adds, “we’ve found that many call centers love this feature since it’s a free and effective tool that allows them to improve company performance.”

In addition, the call recording feature is greatly customizable and easy to use. Every user with a Virtual PBX extension can record all calls automatically or select calls to record manually by simply pushing #9 during a call.  Furthermore, the built-in selective recording architecture allows calls to be recorded for pre-selected extensions and/or departments.

The recordings can be sorted by date/time, length, caller ID, user name, or extension number so they are easy to locate when needed. Recordings can be played back or deleted at the convenience of the extension owner, or downloaded them to a computer for future playback.

Administrators can also decide to control call recording for all users or give each user the ability to manage his or her recordings.

“With the addition of call recording, Virtual PBX extends our leadership in providing clients with the most diverse set of features at the lowest cost,” continues Brashier.  “This announcement gives users the ability to improve their own customer satisfaction and ensure that high priority calls are properly managed.”

This announcement adds to a long list of service enhancements that have been provided by Virtual PBX during 2010. Previous upgrades have included unlimited extensions and more free usage, open VoIP peering, free conferencing, and international numbers. The company promises more to come.

A Mobile Minute with Brian Dally

We decided to take some time this week to introduce one of the newest members of the Bandwidth team, Brian Dally. As our VP of Product Strategy and Mobile Solutions, Brian spends the majority of his time planning Bandwidth’s delivery of the next generation of services and solutions. Previously, he served communications in a variety of global product management and operational roles at high growth technology companies, including Motricity, Openwave and broadband pioneer Excite@Home. He holds an MBA from Harvard Business School and a JD from Harvard Law School. With “Mobile Solutions” in this title, we decided to grill him about Bandwidth’s mobile aspirations.

Sean Rivers - Why would a business voice service provider like Bandwidth be interested in mobile phone service?

Brian Dally - Mobile phones have become critical productivity tools for owners and employees of the small- and medium-sized businesses that Bandwidth serves. We see significant opportunities to help these companies achieve gains in productivity combined with significant cost savings while minimizing the hassle. All of this while enabling businesses to communicate on-the-go.

SR - How do small and medium business take advantage of the mobile technology previously only available to larger corporations?

BD - Smartphones are rapidly becoming the exception that swallows the rule! Just look at the volume of phone calls, emails and web traffic flowing through mobile phones today versus our desktop phones and computers. Fortune 500 companies can afford to integrate solutions into their telecom and IT infrastructure, but for smaller companies, each mobile phone is an island unto itself.  Bandwidth wants to fix that.

SR - With your vast mobility experience, why Bandwidth.com?

BD - I started off my career in mobile by building early generations of embedded micro-web browsers for feature phones, such as Sprint’s LG5350. Back then, mobile was about voice, but as screen sizes and resolutions increased, input methods improved and phones became more powerful - the game changed. Now, it’s all about data - to the point that even voice service is moving that way! Bandwidth is perfectly positioned to create enormous customer value in this market.

SR - What is next for the business mobility industry?

BD - The past several years have yielded an incredible amount of mass adoption of core mobile data technologies. At the same time, some interesting new possibilities have developed for VoIP technology in the mobile context. Just as the [mobile] industry did with mobile email, web browsing and applications, it’s time to separate the mobile VoIP (voice over IP) science projects from the customer-relevant innovations.

SR - What is next for Bandwidth.com?

BD - Suffice to say, @Phonebooth has some really worthwhile mobile tricks up its sleeve.  Stay tuned!

Photo credit: Milica Sekulic

Tags: bandwidth, Bandwidth.com, Brian Dally, Interview, medium businesses, mobile, mobile solutions, Phonebooth, product strategy, small business, SMB, Telecom Advice

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Capacity - Team Bandwidth’s Race Across America Story

After months of planning and intense training, Team Bandwidth took the country by two wheels this last June on their Race Across America (RAAM). Not only did the team finish the 3,004 mile event in 6 days and 3 hours, they also ended up winning the event (4 person team category).

I went along for the ride in the media vehicle and from the footage gathered, I put together an 8-part video series called “Capacity”. The movie is a mixture of story, scenery and intense cycling footage that outlines the highs and lows of crew and riders as they struggle to stay in the front of the pack.

We congratulate riders and crew on their amazing race from Oceanside, California to Annapolis, Maryland, and commemorate the experience through film and photography. See it for yourself….

Capacity - Team Bandwidth’s Race Across America (YouTube)

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Elastix

This is an open source Unified Communications Server software that is offered with IP PBX, email, faxing and collaboration functionality. The Web interface includes capabilities such as Call Center software with predictive dialing. Elastix integrates several software packages each including their own set of features. Elastix adds new interface for controlling and reporting. Elsastix offers Call recording, Voicemail and Voicemail-to-Email functionality, flexible IVR configurable by web interface, voice synthesis support, extension batch tool to generate large number of extensions using CSV files, integrated Echo Canceller. Phone provisioner, configurable via a web interface. It allows configuration of a large number of IP phones in a short time for supported phones.   It features a web interface and includes capabilities such as call center software with predictive dialing. The Elastix functionality is based on open source projects and offers PBX, fax, instant messaging and email functions.

FreeSwitch

FreeSwitch is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media.  FreeSWITCH offers support to various stable telephony platform on which many telephony applications can be developed using a wide range of free tools. It was originally designed and implemented by focus on several design goals including modularity, cross-platform support, scalability and stability. FreeSWITCH offers many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It can also be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.

CallButler

This is a free Windows-based open source PBX, IVR and Auto-Attendant Phone System built on .NET. It is a commercially available product, but is now being put into Open Source. CallButler features Text to Speech, Find Me/Follow Me, Call Forwarding/Transfer, Call Recording, Conferencing, Call Personalization and Unified Messaging, CRM/SFA/ERP/Database Integration.

VoicePulse VoIP Service

VoicePulse for Business offers PBX installer and reseller channel for commercial businesses and wholesalers. The VoicePulse services are compatible with PBX clients like Digium's Asterisk, AsteriskNow, Switchvox, Fonality's PBXtra, trixbox, PBX-in-a-Flash, FreePBX, or FreeSwitch. It is a Asterisk-based systems that allow users to instantly activate new numbers, have active fail overs protection for numbers individually and view inventory of numbers.

CoreDial VoiceAxis 3.0

CoreDial's VoiceAxis 3.0 software platform offers Service Providers and Interconnects to sell, provision, manage and invoice hosted PBX, VoIP, SIP trunking and related services for business and residential customers. This turnkey package is a software package targeted to enterprise-level business and service providers. The open source PBX software has been designed for the  management, billing and provisioning of hosted PBX. It's a one stop management tool that Asterik users can take to, for its ease of use. VoiceAxis management software can be used by medium to large enterprises to implement a cost effective feature rich, scalable Asterisk PBX solution.

How to Follow Team Bandwidth as they Race Across America

Team Bandwidth.com is rocking the Race Across America course. They have already cleared the Rocky Mountains and have a little over 2,000 miles left to go. We know you want all the details you can get, so here are all the ways you can keep up with the cycling action:

Bandwidth.com Facebook Page - This is the hub for all things Team Bandwidth. From here you can see the team’s location via Spot GPS, see Sledge’s awesome camera shots and get updates from the RAAM Crew.Bandwidth on Twitter - You can learn about the adventure on @Bandwidth Twitter account, as well. The official hashtag for our team is #BandwidthRAAM. Also a quick search for #RAAM2010 will give you a live feed of everything people are saying about this year’s Race Across America.Bandwidth on Flickr - View all of Sledge’s camera handiwork. Here you can see all his great images that he shot to document the RAAM journey.Bandwidth.com over the Phone - That’s right, we called in a favor from @Phonebooth. Fans can call 919-442-8920 to hear updates from the riders as they peddle their way across the country towards the finish line in Maryland.Bandwidth.com’s RAAM Challenge - We are literaly racing our RAAM team across the country. On the Intranet, you can view the total miles accumulated by Bandwidth’s RAAM Team as compared to the Home Team. Also, Bandwidth Home Team, don’t forget to fill out your workouts. The first team to accumulate 3,004 miles gets Friday, July 2nd off!

Hooray for Small Businesses Driving the Economy

In honor of National Small Business Week (May 23-29, 2010), we would like to celebrate our 6,000-plus small business customers.

Small businesses drive the economy by spearheading innovation, creating jobs and setting up shop in the competitive marketplace. These entrepreneurs make business decisions to simplify their lives and others. As a whole, the small business community advances businesses of all sizes toward bigger and brighter fiscal years. It’s a great time for the estimated 27 million U.S. small businesses to be steering the wheel.

We, here at Bandwidth.com, would like to do our part to help make sure your needs are met. Please take a moment to participate in our short poll. As a small business, we want to know what you consider to be your biggest communication challenge.

A few interesting SXSW panel proposals from Bandwidth

If you’ve ever attended SXSW in Austin, TX… you know that there is a great amount of diversity in the panels submitted and presented. We felt like voice was an area that had been lacking and submitted a few panels this year. If any jump out to you, please follow the link and take a minute or two to submit a vote. Thanks for your support!

We Messed Up – Why Telecom Is All Wrong - click to vote

Users and carriers battling it out over dropped calls and bad service while networks explode all around us. What would the telecom world look like if everyone simply got along? In many ways we’ve moved backwards. All this technology should be better than it is, but why are there still such problems? It’s a strange thing because the technology is here now. What stands in its way is an archaic method of business that has everyone unhappy. Let’s move past today and into the future of all things phone. In this panel you’ll hear from leading innovators in the field of telecom where they’ll explore a world in which customers are happy with their carriers, telephony apps abound and networks are stable. They’ll describe what that looks like and why it’s still possible.

Endurance and Entrepreneurship - click to vote

Where does mental motivation meet physical motivation in the life of an entrepreneur? The qualities best suited for running a company are often the same as any physical fitness training – hard work, dedication, competitive spirit and an all-or-nothing attitude. How does living a healthy, proactive lifestyle make you a better founder, CEO and leader? In this panel we’ll speak with some of today’s most fit CEOs about how physical challenges to their mind and body translate into their company culture and workplace. And more importantly, how being physically and mentally fit is the best way to win out over the competition.

Speak to Me: Plugging Voice into Your Apps - click to vote

Voice apps are exploding and developing at a record pace. The world of tomorrow will mean you interacting with your phone more as a concierge than simply a way to communicate. With invention in areas such as telephony apps, dial tones and whisper technology, come hear what’s next. This panel of leading experts and innovators will take you behind the red curtain to explore the mind-boggling labs and phone projects in development today that you didn’t think were possible. Targeted at a more technical developer crowd, the panel will show examples of apps that are plugged-in to the latest in voice APIs.

You Can’t Do Dat! Innovating Under FCC Regulation - click to vote

Starting up is hard enough already, but what if you are setting up shop in a heavily regulated industry? These can be some of the most lucrative and the most tricky ventures. So, how can entrepreneurs and their technology-based innovation deal with a rulebook constantly getting in their way? In this panel, we’ll explore real-world stories about how startups and innovators succeeded in an environment that was heavily regulated by the FCC. Experts from the telecom and VoIP world will share their advice on how your startup and your technology can handle the red tape and even use it to your advantage. Are you under-innovating because of a fear of regulation? When it seems like all is against you, it may simply be the final hurdle before success.

sipXecs Enterprise SIP PBX

sipXecs is an Enterprise SIP PBX that comes complete with voice mail and auto-attendant. It can also be used as a high performance enterprise toll-bypass SIP router. sipXecs combines all common calling features, XML-based SIP call routing, Web-based configuration, and integrated management and configuration of the PBX and attached phones and gateways. It is a modular server-based solution that does not require any additional hardware, as it interoperates with any SIP compliant gateway, phone, or application.

Digium Open Source Telephony

About Digium
Digium®, Inc., the Asterisk® Company, created, owns and is the innovative force behind Asterisk, the most widely used open source telephony software. Since its founding in 1999, Digium has become the open source alternative to proprietary communication providers, with offerings that cost as much as 80 percent less. Digium offers Asterisk software free to the open source community and offers Switchvox, Digium's Unified Communications solution to power a broad family of products for small, medium and large businesses. The company's product line includes a wide range of telephony hardware and software to enable resellers and customers to implement turnkey VoIP systems or to design their own custom communications solutions. At Digium, we're changing the way businesses communicate.

A Peek at Cloud Telephony: SIPfoundry’s sipXecs

A recent article posted in The Technoverse Blog (in Telecom Patchboard)

My curiosity got the better of me.  While I’m completely content to use turn-key cloud telephony–OnSIP, in my case—the lure of DIY telecom is sometimes too enticing to resist.

This led me to SIPfoundry’s sipXecs, an open-source PBX that many are using instead of an on-premises metal-based solution.

SIPfoundry has grand goals for open VoIP solutions. They are an independent non-profit that hopes to promote “free and unencumbered” telephony. Which is another way of saying their sipXecs PBX software is 100% standards based. So if enough companies, small and large, install sipXecs on their servers, we can all communicate via SIP over the Internet and not pay a dime in per minute charges.

I thought I’d experiment with sipXecs to see what all the shouting was about.

I am not part of a corporate structure with spare LINUX servers on tap.  I did what a lot of business that need on-the-fly access to  data center servers are trying: I grabbed a virtual server from Amazon’s Elastic Computing Cloud or EC2.

Did I also mention I’m not really an IT person? Certainly sipXecs requires a very tech savvy person to install and maintain. I knew about EC2 from a previous writing assignment, and I was after all a former UNIX developer.

Configuring an auto-attendant with sipXecs

Just enough background to get me  into trouble.

The first speedbump I ran into was learning enough about EC2 to grab an appropriate virtual instances—in my case I was looking for Red Hat’s Centos version 5 OS—from their  data center in the clouds. There are references at the end to explain how to access EC2: the key tool being Amazon Web Services or AWS console.

EC2 and AWS are not terribly difficult to comprehend and work with, but there are subtleties with private and public IP addresses and DNS—some of which is still stumping me.

My goal was modest: just to bring sipXecs up and experiment with its browser-based interface. If I could get a SIP endpoint connected, I would consider that just gravy.

I forged ahead with my foundry.

The documentation on their Wiki explains how to install the software—you’ll need to learn a little about the yum software installation utility.  So … once the sipXec is installed, you then  configure this soft PBX using their sipxecs-setup command.

After a little trial-and-error, I got sipXecs on-line and then scooted into the browser interface.

From what I can tell, this thing looks like a real PBX:ACD, auto-attendant, conferencing, hunt groups, intercom (automatic answer), along with support for lots of SIP phones (Cisco, Avaya, Linksys, Polycom, Audiocodes, …) and gateway integrations to the TDM world.

Overall, I am impressed. Quibbles: the responsiveness of the Amazon virtual OS instance I’m using is sluggish, but I didn’t pay for anything very powerful.

Yes, did try to get my X-Lite softphone to connect;unfortunately that involves a level of DNS prowess that I don’t possess at this point. I hope to have a resolution soon enough and should have another post on how sipXecs plays with endpoints.

I was on do-it-yourself-mission in this post but that shouldn’t take away from the fact that sipXecs is a serious product for large companies.  For the enterprise,  a new player, eZuce, has recent stepped in to provide corporate-level support.

Even with pay-for-support model, I believe that sipXecs is very competitive proposition versus on-premise hardware. Check out the eZuce site for my information. read more....

Sneak Peek: Upcoming openACD release

As many of you know we have been working diligently on a new contact center ACD solution called openACD to replace the currently existing ACD application in openUC / sipXecs.  We are making rapid progress and it is time to give you a sneak peek into what is coming.  There is a lot of excitement about openACD as it holds the promise of becoming a highly flexible, scalable and resilient contact center solution.  openACD is one of these next generation contact center solutions that is media agnostic and allows queuing lots of things, including basic calls, voicemail messages, email and at some point instant messages (IM) or even FAX.   It is skills based and offers priority queuing implementing rules that look at different criteria for incoming messages.

Here are some of the highlights:

- Contact center ACD solution with skills based routing, priority and unified queuing, openUC / sipXecs integration for Web administration, supervisor interface for managing agents and call flow, and detailed CDR recording.

- Agents now have the flexibility to transfer calls, using an agent Web UI, in a variety of ways; something that was not possible with our old ACD. Agents can transfer a call back into queue, or into another queue, or to another agent, or they can do an attended warm transfer to an external number.  All these transfers can be tracked for historic reporting.

- Callers can press star '*' while in queue and leave a voicemail instead of staying in queue.  The voicemail message then stays in queue and is presented to an agent.  The agent can then decide to either call the caller back or respond in a different way if the caller's contact information is available.  The voicemail is treated with lower priority and put at the end of the queue by default, but that could be configurable at some point so as to maintain position in the queue.

- openACD can queue email and such incoming email messages are presented to agents over a Web UI.  As with calls there is a wrap-up time that applies to responding to an email.

- Supervisors have a lot of flexibility and different ways to intervene, including transfer calls to a particular agent, send them to voicemail, listen in on a call including whisper and barge.  If emails are in queue supervisors can view the email including mark it as spam.  Call recording is supported, although we will have to improve on recording granularity a bit.  The supervisors can also move agents between profiles to balance load across clients/departments during a call volume spike. There's also support for the supervisor to record an outage message to be played as part of the IVR.

- Agents can be remote either as remote workers connected to the system over SIP, or using an external phone connected to the PSTN.  Care needs to be taken so that calls do not ring out to voicemail systems outside the call center environment.

One of the most exciting aspects of openACD is the possibility to achieve significant scale and redundancy.  Some more work will be required in this area, but here is the basic concept:  Several openACD nodes can be setup as part of an openUC / sipXecs system, all centrally managed and part of the cluster.  Queues can span more than one node and accept calls from any node and deliver to an agent on any node.  Queues are global to the cluster by default but can also be split and then merged again.  This is useful to establish redundancy in case connectivity is lost between nodes or to scale a queue to include several servers.  When a queue is split it continues to operate with its own version of the queue containing only the calls present on that node.  The queue can then be merged again later, combining the calls.

Early performance and scalability tests are very promising with a range of different results.  We will publish results once we are a bit further along in testing.  If you have results to share let us know.

openACD is available in our developer builds and you can participate in the testing effort.  A first official release will be included in the openUC / sipXecs 4.6 release.  If you are a company with a need for a cost-effective contact center and would like to work with us, now is the time to influence the direction we are taking with this. 

And finally, I'd like to thank everyone involved, in particular the openACD team Andrew and Micah, the KGB team without whom this would not have been possible, as well as the team here at eZuce.   We are very excited about this new capability added to openUC / sipXecs and we are looking forward to sharing more details soon.

SIPfoundry and the University Community Announce Unified Communications Initiative

NEWBURYPORT, MA--(Marketwire - September 29, 2010) - Today SIPfoundry launches its EDU initiative, a comprehensive program designed to bring lower cost and modern communications to universities, colleges and school districts. As a proven replacement for legacy IP-PBX systems and as an open alternative to an all Microsoft infrastructure, the SIPfoundry sipXecs solution brings a robust all software solution with a sophisticated user experience to the higher education community.

Colorado State University (CSU) and Cedarville University are the first full members in this joint effort to transition from costly, limiting, legacy IP-PBXs and build 21st century communications that provide a rich, connected experience for faculty, staff, students and alumni. As investing members, CSU and Cedarville represent the higher education community to drive the future direction and define the requirements for an open, interoperable communication solution.

News Facts

The SIPfoundry EDU program is designed to inspire innovation for the open source UC solution of the future based upon the sipXecs open source code, the largest and most comprehensive open source effort to build unified communications using the Session Initiation Protocol (SIP).SIPfoundry embraces a 'For EDU - By EDU' approach that ensures higher education institutions save significant money and it gives them direct influence over the future direction and roadmap of the open source development.The SIPfoundry website has been updated to make it easy for members to collaborate, share best practices and leverage open source applications.In addition to investing in SIPfoundry, Patrick Burns, VP for Information Technology and Dean of Libraries at CSU has joined the SIPfoundry governing board of directors. You Tube: Video Testimonial from Patrick Burns at Colorado State University

Supporting Quote
"Our campus selected sipXecs because we wanted to invest in a product that allowed us to save money while facilitating collaboration with our peers," said David Rotman, Associate VP for Technology CIO, Cedarville College. "We are confident that this program will enhance our ability to communicate and is well suited to meet the learning, research and academic needs of our university and others. We are especially pleased to be involved in a project with significant user-community involvement."

Supporting Resources
SIPfoundry Homepage
SIPfoundry and eZuce EDU Program
SIPfoundry Twitter Page 
SIPfoundry Facebook Page

About SIPfoundry 
SIPfoundry, the leading collaborative open source community dedicated to unified communications and founded in 2004 by Dr. Martin Steinmann and Jerry Stabile as a not-for-profit organization, promotes and advances Session Initiation protocol (SIP)-related open source projects. eZuce Inc., established in 2010 by the creators of SIPfoundry, is the commercial entity that delivers open enterprise communications solutions and support based on the standards established by the SIPfoundry community.

BroadVoice Internet Phone Service

BroadVoice™ Internet phone service allows residential and business customers to use their cable modem, DSL modem, or other BroadBand Internet connection to make and receive Voice over IP (VoIP) phone calls using an ordinary touchtone telephone. Bring Your Own Device™ (BYOD™) plans allow customers to connect their own SIP devices, including IP phones, softphones, and Asterisk PBXs. BroadVoice utilizes our SecureSIP™ technology to ensure accurate connectivity throughout the user experience. SmartSIP™ technology by BroadVoice is used to optimize the routing of network voice traffic, provides the best possible quality voice transmission for each customer's phone device, and automatically configures BroadVoice Authorized BYOD™ devices

Vocalocity Review

The VoIP provider Vocalocity say that it markets only to small businesses, and that almost all of its customers are companies with 50 employees or fewer. As a result, Vocalocity claims to understand the small business market and offer products that work best for small businesses. The company claims in its website that using a Vocalocity system can reduce your costs by 50-80% over a traditional on-premise PBX system.

How does it work? By delivering all of your telephone services over the internet instead of the copper lines of the phones company. All you actually need are some phones and a broadband internet connection. Vocalocity owns all the heavy-duty equipment like routers and switches and takes care of maintaining and upgrading th software.

You just pick up your phone like you always have and make your calls. 

Vocalocity’s most popular plan costs just $39.95 per month per person for unlimited inbound and outbound calls, and includes a whole laundry list of features including: caller ID, call waiting, forwarding and holding, 911, 411, conferencing, do not disturb, voicemail and detailed call logging, to name a few.

And because it all runs on the internet, you can access all your phone features from and computer and phone with an internet connection. So your employees in remote locations or even with cell phones can access all the same features. Plus, you get an internet portal; to help you add extensions, change rules for night lines and forwarding, and to manage every other aspect of your account.

There is no limit to the number of callers you can have, save for bandwidth considerations. Figure that you will need about 64 kbps of bandwidth per simultaneous call. Check with your internet service provider to see how much bandwidth you have available in your connection. If you will be making a large number of simultaneous calls, you might need to increase your bandwidth.

Vocalocity also offers a lower-cost plan. The metered plan is just $14.95 per month plus 3 cents per minute for calls. Yo might use this in a conference room or a visitor lobby, where call are mostly internal.

With all the above plans, international calls are extra, with prices based on the country you are calling and the phone system within that country. Prices run from about a nickel (France to as much as $1 (Belarus) per minute.

Other add-ons, which you would purchase a-la-carte, include additional local numbers, toll-free numbers, virtual local numbers ( a local number in a city where you are not located, but perhaps your customers are), paperless faxing call groups and more.

In addition to traditional customer service, the company offers webinars, white papers, case stories and other resources such as FAQs.

Vocalocity is a privately held company headquartered in Atlanta, Georgia. They can be reached at 1-877-862-2562 or at vocalocity.com. Customer service can be reached Monday through Friday from 9 am to 9 pm (EST) at 866-499-9474.

Mitel PBX Phone Systems

Mitel’s offerings to the voice communications market seem oriented to the large user and the service provider with smaller customers, rather than the small- or medium-sized business markets. By providing both equipment and negotiated rates with U.S. signal carriers, Mitel offers to be your communications partner.

Hardware / Infrastructure – The company’s flagship product line – the model 3300 IP Communications Platform – is scalable from 10 to 65,000 users – -that’s right – 65,000 users. It is a full-function PBX system with embedded capabilities including unified messaging and auto-attendants. It is designed to run in most Local Area Network (LAN) and Wide Area Network (WAN) environments, and supports IP phones, wireless phones and audio conferencing gear.

Other “infrastructure” offerings include set-ups for up to 52, 250, 600 and 1,000 users. A further offering enables the 3300 series to run on Sun Microsystems hardware.

Software – the Communications Director program — is designed to run on Mitel’s proprietary hardware or on industry-standard servers from third-party suppliers. The company also offers a Multi-Instance version in which multiple copies can be run on a server or servers to offer the same functionality to a large user or – in the case of a service provider – multiple small and medium users.

Management and Reporting – Mitel offers programs aimed at simplifying and improving the management, reporting and troubleshooting of Mitel networks. Enterprise manager includes a suite of tolls for configuring and controlling even enterprise-wide systems from a single portal, reducing costs and streamlining administration.

Another program offers remote management functions, enabling outside engineers to diagnose and troubleshoot system problems even from a faraway location, reducing the need – and expense – of on-site visits.

Phones and Accessories — Mitel offers dozens of of telephones and accessories to go with its systems. Ranging from basic IP-enabled desktop phones to high-end multi-line speakerphones and special-use units such as the model 5560 IPT which designed to withstand the rigors of the stock-exchange trading floor. Accessories and peripherals include everything from headsets and stands to interfaces for analog phones, programmable key modules softphones (a phone run on a PC) and digital handsets.

 

Applications – Mitel’s applications suite offers a broad range of programs designed to add increased functionality to the company’s systems. Applications include Business Dashboard for telephone-system reporting and management, Customer Service Manager, Unified Messaging to integrates voice with other communications, audio and web conferencing, teleworkers and a mobile program for unified communications.

Wireless — Another area in which Mitel has a broad range of offerings is in wireless products. It offers phones, gateways, wireless IP phones, phones designed to work in the ubiquitous 802.11b wi-fi environment (including PDA-based softphones).

Finally, Mitel offers a variety of services to complement it’s hardware and software offerings. Choices include everything from systems design and engineering services to hosted solution to completely outsourced management of your phone system.

Mitel has even negotiated deals with a number of ISPs around the country for reduced cost signal carriage, and may be able to offer competitive rates for phone, Internet, data transport and other messaging services.

Mitel sells through a network of reseller partners. Given the large numbers of both resellers and possible combinations of software, hardware and services, it is impossible to say anything meaningful here about pricing.

According to Techcrunch.com, Mitel, a privately-held company with annual sales of almost $750 million and more than 2,000 employees, is in the midst of an IPO filing in Canada worth an estimated USD$230 million.

The company is headquartered in Kanata, Ontario, Canada and can be reached there at 613-592-2122. U.S. operations are based in Chandler, Arizona, and can be reached at 480-961-9000. Customer Service is open from 8 am to 6 pm EST and can be reached at 1-800-267-6244.

TelaSIP SIP Overview

Telasip is an ITSP – Internet Telephone Service Provider – that offers Session Initiation Protocol (SIP)-based Internet telephone service to residential and business customers. With an SIP plan, the provider replaces the phone company’s twisted-copper wires with the Internet, where it is much cheaper to send data.

The company offers three residential plans:

VoIP Plus – $14.95 per month for unlimited incoming plus 500 minutes to the US and CanadaVoIP Premium – $24.95 per month for unlimited incoming plus 1500 minutes o the US48 and CanadaVoIP World – $34.95 per month for unlimited incoming plus 1500 minutes to the US48, Canada and selected international locations.

All residential programs include the ability to have 2 concurrent calls, and include call waiting, caller ID, three-way calling, voicemail, anonymous call rejection and caller-ID block plus emergency calling services. Virtual numbers and enhanced 911 are available as options for additional fees, as is international calling, with rates depending on the country called (a full list is available on the company website).

There are various ways you can purchase business VoIP services from Telasip. While the company website is very short on details (they are planing on re-designing it this year), a customer service representative was very helpful in filing in the blanks. Available plans include:

A by-the-minute plan (1.1¢/minute for incoming and 2.5¢/minute for outgoing) with a minimum of $40.
A basic 3,000-outgoing-minutes plan with unlimited incoming calls, voice-mail to e-mail and fax-to-e-mail features for $44.95 per month. Additional minutes cost 2.5¢ each.
Hosted flat-rate plans with 12 or 25 extensions for $249.94 and $399.95 per month respectively
Enterprise plans beginning at 24 ports with up to 60,000 minutes starting at $699.95 per month

Business plans include such features as call waiting and forwarding and caller ID.

The company also offers per minute, per port and carrier-level wholesale pricing on both inbound and outbound calling services.

As described in the company website, the company is as expanding its network to improve service levels and call quality. It has plans top build new facilities in Salt Lake City, Atlanta and Boston tp supplement the eight gateways it currently operates.

Telasip, which also does business as Epraxys, Inc., is a privately-held company based in Montgomery Village, Maryland. They can be reached at 240-396-1450.

Vitelity SIP Trunking

When you use SIP trunking, you no longer make your phone calls through the twisted-pair copper-wires of the phone company.  You connect your PBX to the Internet.  Your Internet Telephony Service Provider, in this case a company called Vitelity, takes calls made on your office PBX and makes them compatible with VoIP technologies, routes them as far as they can using the Internet, and then, if necessary, back onto the POTS (Plain Old Telephone System, also known as the PSTN, or Public Switched Telephone Network) if needed, to complete the call.

People generally switch to VoIP and SIP trunking because it can save them money on their phone service versus running calls over the regular phone system.  Many companies report as much as a 70% drop in costs. There will be some up front costs because you will need specialized hardware and software to use VoIP, but you should be able to recoup your costs in short order.

Vitelity Link Service is a no-commitment, pay-as-you-go service for small and medium businesses that starts at just $35 per month for the “dial-tone”. A local phone number in most areas of the U.S., with unlimited incoming calls from the “lower 48”, costs $7.95 per month with a $1 activation fee and outgoing call terminations cost 1.44¢ per minute for calls terminating anywhere in the U.S.  Custom toll-free and V-fax numbers are also available at an extra charge. Local and toll-free numbers can also be ported over.

With phone company service, you calculated how many concurrent calls you wanted to be able to support, and got that many incoming lines.  With SIP trunking, the limiting factor isn’t lines, it is bandwidth.  An average call uses around 65kbps (65,000 bits per second). SIP trunks carry VoIP signals along with your Internet traffic, so it is a good idea to work with your ITSP to determine how many calls you can support with your available bandwidth.

Vitelity supports SIP and IAX protocols and supports several compression/decompression (codec) protocols.  In larger markets, Vitelity can bring in to your office a T-1 line (1.544 megabits/second) or enough to carry 23 average phone calls at once.

A variety of features are available through Vitelity, all on an a la carte pricing model, including:

Enhanced 411 (99¢ per call)Outbound caller ID name ($10 per number)Call forwarding (2.8¢ – 3.5¢ per minute)Enhanced 911 ($1.49/number)Virtual PRI – If you need large numbers of inbound lines (pay by the line, not the minute)Voicemail – up to 1000 mailboxesCustomer portal – An Internet-based portal to help manager your account and calls, produce usage reports, add users and more

In a review of several chat forums and review sites dedicated to this market, Vitelity earned very good marks for customer service and support.  With few exceptions, the company was praised for answering customer questions quickly and completely.  Even on those sites dedicated to complaints, Vitelity fared well. Customers felt the company gave excellent service and delivered on what it promised.

Vitelity is part of U.S. National Telecom, a Miami, Florida company traded over the counter under the symbol USNL.  Vitelity, based in Denver, Colorado, can be reached at 1-720-257-5400 or 888-89VITEL (888-898-4835).  Customer support is available by phone from 9 am to 5:30 pm MST.

Digium SwitchVox Systems

Imagine when a phone call comes in, you can look at your computer screen and the Switchvox “switchboard”  program knows who is calling, shows you their name, number and a map of where they are calling from and even brings up an automatic “Google”  page about the caller.  If you have SalesForce, you ‘ll also get that program’s file about the caller plus any comments left on the system after the last time that person called – all before you even pick up your phone.   

You can then decide to answer, transfer to another extension, send it to voicemail or record the call – or finally pick up the phone and answer!  Add to that the fact that you can do so from any IP enabled phone on your network – no matter where it is located – and you begin to see the design philosophy of the company’s premier proprietary product, the Switchvox. 

Come for the hardware lineup……stay for the awesome dance song

The Switchvox phone system comes in three sizes:

Concurrent recordings/conference call users5 recording5 conference callers10 recording15 conference callers20 recording30 conference callersRAID controller, mirrored drive, redundant power supply & optional cold spare failover

Unlike most suppliers, Digium does offer a free trial version of Switchvox.  It is downloadable from the company website and it supports up to 15 users.  But be careful – only download to a computer you don’t use for something else because the software you download will wipe your hard drive.

Support Plans

Silver – $55/user first year then $11/ userActivation plus unlimited e-mail supportGold -  $77/user first year then $17/userActivation plus unlimited e-mail and phone support (during normal business hours of 6 am to 6 pm PST)Platinum – $110/user first year then $28/userActivation plus unlimited e-mail and phone support (during normal business hours of 6 am to 6 pm PST) plus 5 after hours incidents

Beyond the above basic components, options of all sorts are available on an a la carte basis, including:

Analog phone linesEquipment maintenance and warranty plansTelephones – Mostly Polycom Soundpoint phones (from $174 to $549)Conference phonesT-1 lines

Switchvox components and systems are available direct from the manufacturer or through a variety of third-party resellers. 

The company is also the creator of “Asterisk” software, an open source program that can turn an ordinary computer into a communications portal or server.  In combination with Linux and other open-source third-party programs you can assemble a complete communications system from open-sources (non-licensed) programs. 

Both Asterisk and Switchvox are products of Digium, Inc., a privately-held company located in Huntsville, Alabama.  Digium can be contacted at 256-428-6000 or 1-877-DIGIUM1.  Customer support is available from 7 am to 8 pm M-F at 1-258-428-6000.  The company website is www. Digium.com

AireSpring Data/SIP Review

AireSpring offers a broad range of telecommunications services for customers ranging from small businesses with just a few lines all the way up to full enterprise coverage.  Product offerings extend to not just local and long distance voice calls, but also include high-volume data applications: 

High-volume local and long distanceVoIP / SIP (local and long-distance)High-bandwidth data (up to 10 Mbps)Conference CallingMultiple-location and enterprise networks 

If you have a standard PBX requiring a TDM handoff, AireSpring’s offers Dynamic X,  a T-1 line which brings together voice and data over the same line.  Instead of being forced to allocate fixed percentages of your bandwidth to either voice or data,  Dynamic X, as it name suggests, dynamically allocates your bandwidth to voice or data (across up to 16 IP addresses), giving voice calls priority when needed. 

If your PBX is IP-enabled, AireSpring offers traditional SIP trunking.  Get rid of your copper phone lines and PRI (T-1 service) completely and go exclusively over the Internet for your phone service.  Because calls go out over the Internet, AireSpring can use least-cost routing to cut your phone bills.  Trunks start at just $8 each per month.   

As with all SIP-based systems, available bandwidth will be the primary factor determining how many simultaneous calls your phone system will be able to handle.  Make sure you have enough to handle your calls plus your anticipated data needs.  

AireSpring also offers virtual DIDs (local phone numbers) as a way to reduce your use of expensive 800 numbers.  Let’s say you are based in New York, but you get a lot of 800 number calls from Chicago.  Drop the 800 number and replace it with a virtual DID in Chicago for 50¢ per month (including usage).  It works like a local number for your customers but costs you much less. 

SIP trunking is entirely Internet based, so all the features available in your headquarters are available to your entire network. You will no longer be location-dependent for your phone service. Callers won’t see a difference between an extension in your office and an at-home worker or someone calling from on the road.  And all calls between and among network users will be free. 

If you have an existing data or voice network running across multiple carriers, AireSpring’s offers a Virtual Private Network (VPN) offering Multiple Protocol Label Switching (MPLS) a technology which enables packets to be switched between carriers regardless of the transmission protocols involved.  Run all your  traffic -voice, video and data – over a single network. 

Pricing for the company’s varying product offerings is based on a number of factors, including: 

The number of concurrent calls you want your system to be able to handleThe combination you want of local, long distance, data and teleconferencingThe types of equipment you have and its locations. 

 AireSpring has negotiated deals with Tier-1 carriers and currently covers 95% of the United States.

According to the company website, the network is designed to accommodate large-volume and high-capacity users.  Partner companies include such well-known names as Verizon, Global Crossing and Qwest. 

AireSpring, is a privately-held company located in Van Nuys, California, and can be reached at 1-800-825-1055.  For support, call 888-389-2899 or visit www.airespring.com.

VoicePulse SIP Overview

VoicePulse is a supplier of VoIP Telephony services aimed mostly at the residential (SOHO) and small business market, with the ability to serve larger customers and resellers as needed.  VoicePulse services use SIP (Session Initiation Protocol) technology that routes calls over the Internet instead of traditional phone company copper lines.

VoicePulse is primarily a supplier of services.  Although it does provide routers for certain service plans, VoicePulse customers generally supply their own PBX hardware and phones.  VoicePulse’s phone service includes a comprehensive range of features, included three-way calling, caller-ID with name, call waiting with caller-ID name, call return (*69) and call transfer. 

Other features enable you to block selected calls (such as telemarketers or calls without caller-ID) and to filter which calls get through and which ones won’t.  Also included are additional features such as speed-dialing and distinctive rings.

There are five basic service plans listed on the company’s website: 

Outgoing Long Distance (50 US States) call minutesAs low as 1.5¢ / Minute (Lower 48 states only)Outgoing Local and Regional US Calls*Cost/Min. Varies (see website)Cost/Min. Varies (see website)

*See website for breakdown of what comprises local and regional calls 

** Customer is responsible for providing SIP-compliant equipment.  SIP trunking typically comes with enough bandwidth for four simultaneous calls.  Additional simultaneous call c\apacity costs $20 per call.

As with most suppliers of VoIP, there is an entire laundry list of services and features available at an extra cost.  With VoicePulse, that list of extras includes phone numbers, activations, conference-calling capabilities (free with Business Unlimited only), virtual numbers (local numbers in another area code) and more.  Check the website carefully to be sure what will cost extra and what is included.

VoicePulse also offers custom “solutions” for such applications as large offices (20+ users), call centers, colleges and universities, and a wide variety of wholesalers / resellers of VoIP services.

VoicePulse is a privately-held company headquartered in North Brunswick, New Jersey.  They can be reached at 732-339-5100.  Customer service can be reached at the same number on weekdays between 9 am and 5 pm EST.

BroadVox SIP Overview

Broadvox is a supplier of low-cost SIP telephone service, which saves you money by routing calls over the Internet instead of over the Public Switched Telephone Network (PSTN), also known as the phone company. By using its own, proprietary network instead of the phone company’s twisted-copper wires, you get much higher quality calls at a lower price.

You can gradually migrate your phone service from old-style Time Division Multiplexing (TDM) technology to the newer digital Session Initiation Protocol (SIP) used in Broadvox’s Voice over Internet Protocol (VoIP) systems. In this new method, your phone call gets converted into data packets, and it travels the Internet the same way as a YouTube video or a spreadsheet.

Broadvox’s basic product is called Go!Anywhere, and it allows unlimited local and long distance

calling, with what the company says will be a roughly 70% savings over the traditional telephone company. You can use your existing equipment or get an Internet-Enabled PBX system. The basic service costs $35 per concurrent call session, with a minimum of 3 sessions required. Prices drop for all services if you contract for more than 1 year of service. Incoming phone service is available for an additional fee, in increments of 5,000 minutes, starting at a little over 3 cents per minute. Or for 3.5 cents per minute a la carte.

The company also offers two variations of the program, one called Go!Local, which features unlimited local calling with the option of adding long-distance. Go!Domestic is a plan for call or contact centers that use T-1 lines. It eliminates the local component of incoming calls, reducing the cost per minute.

To all three of the above programs, the program offers the ability to add extras, such as enhanced local numbers, with which you can customize the outgoing caller ID information, virtual numbers, directory listings and local number portability.

Finally, Broadvox offers wholesale call origination and termination services for resellers.

The company has an extensive group of partner companies with which it works to provide the best possible level of service. Hardware partners include such well-known names as Alcatel / Lucent and Cisco. It also works with more than 35 vendors of IP phones, PBXs and other equipment, to be sure that their equipment is compatible with its networks.

Broadvox is a privately-held company headquartered in Dallas, Texas with it Network Operations Center in Cleveland, Ohio. It can be reached in Dallas at 213-646-8000 and in Cleveland at 216-373-4600.