Save $100 When You Buy a GN Netcom Headset Handset Combo

April 15, 2008 by Garrett Smith

Another great Voip Supply Promo this week!

This week VoIP Supply has a great promotion featuring GN Netcom products. You can save $100 when purchasing a GN9330 and GN1000 combo. The GN 1000 Remote Handset Lifter answers and ends calls remotely, allowing you to roam free around the office. The GN9330 Wireless Headset is the perfect pairing and provides noise cancelling sound with a lifter. It is also WiFi-friendly. Your voice will always be transmitted clearly, even as you are moving through a noisy area. The headset allows a nine-hour talk time without recharging, ensuring you are always ready for every call.

So make sure to check out VoIP Supply, because these essential VoIP items won’t last long. Get a great deal by saving $100 when purchasing this great combo.

Where’s VoIP Going? A Look At VoIP Industry Trends

April 13, 2008 by Garrett Smith

Where is the VoIP industry going?

Of late, it is the question I field most often.

I am not sure if it is because as the market mature, seeing what’s next is increasingly important, or if it is because as the marketplace matures, what worked yesterday is no longer working today (or will work tomorrow). Whatever the case may be, I thought it would be fitting to address where the VoIP industry is going, in a short and sweet high-level digest.

VoIP, in general

The buzz surrounding VoIP has worn off. Overall interest in the VoIP technology is trending downward due to this, although the market is still growing. People have wised up to the fact that VoIP is just another transport mechanism for voice and therefore what is currently hot is the applications and business enhancements that come as a result of using IP for voice transport. The first adopter’s have come and went. This new wave of interested parties are no longer looking to just save a buck, but in almost all cases, is looking at VoIP as a way to “improve” or solve a problem. If you are a VAR, Manufacturer or Service Provider, go there.

VoIP Service

Consolidation, bundled offerings and tighter margins are the name of the game in the service provider marketplace. Over the past few years, hundreds of providers have sprung up and many are now fighting for their lives. Without control of the pipe or a true differentiator, many are now looking to merge or get acquired by other providers. For those who were lucky enough to see that survival could come in the form of bundling multiple services, the trend will continue as getting “multiple services delivered through one pipe from one provider” is the ultimate goal for many service providers. Expect this pursuit and these offerings to continue to come pouring into the marketplace. It won’t be easy though, many, if not most, of the service providers who have entered the marketplace over the last few years, have aided the race to zero and as more struggle to survive, I expect service providers to continue to operate on thinner and thinner margins. Innovate, collaborate or die.

VoIP Hardware

There is not much new here, except in the handset space. The latest here is a move towards pushing applications down to the handsets, applications that improve or make business processes more efficient. While these applications have not yet hit mass adoption by end users, manufacturers are spending big (and talking big) when it comes to this new functionality. In addition to applications, as the pipes get cheaper and larger, more and more vendors (such as Polycom) are creating handsets that can handle larger codec’s that will increase call quality and better the user experience. HD Voice will be big, soon.

Open Source

Open source companies are growing up and therefore there has been an overlaying theme of monetization occurring. Monetization of the technology, the community, the websites and of course the ecosystem. This shouldn’t be viewed as a bad thing, just because these companies are open source does not necessarily mean they should not be pursuing revenues and profitability. They are in business to make money, as is everyone. If your business was impacted as a result of these efforts, should have had the foresight to see where they were going; it was inevitable.

Another side trend, in open source telephony, is the movement towards the “appliance.” I have been talking to open source companies in this space about an appliance based approach since 2005, when a conversation with a small business owner lead me to the thought that most people want a phone system, to look like a phone system (whatever that is) not a PC or a server. I am not crediting myself with the idea, but of late, everyone has their own appliance…and from what I have seen it has been could for their business (and their VAR’s). I don’t see too many more appliances hitting the street as the market is saturated, but I do see the appliances continuing to evolve and becoming a selling point for open source companies when recruiting resellers.

This was pretty brief and honestly, I could go on for hours, but I won’t. What I would like to do though, is open up the floor for others. Have a trend you would like to discuss? Leave a comment below!

FOIP-Fax Over IP Solution

April 11, 2008 by Arthur Miller

Rockbochs introducing Faxxbochs

RockBochs, Inc, creators of the popular Trixbox / Asterisk server Phonebochs, announced earlier this month its Fax over IP Solution, Faxxbochs.

About the Faxxbochs

“A truly reliable Fax over IP solution; This is not a simple ‘E-Fax’ service, but rather a solution that leverages your existing internet connection to provide a cost effective way to transmit and receive paper fax, provide fax to email functionality, automatic archival, with a full web-based management interface.” states Chad Behling, owner of RockBochs, Inc.

Read more about the offerings here:

Faxxbochs

April 2008 New Product Roundup

April 10, 2008 by Garrett Smith

Von and Voicecon Industry Shows recap

The shows have brought new products to the market and the foloowing is a brief recap on the new offerings rolling out.
Grandstream

Grandstream

Brookline, Mass.-based Grandstream Communications has thrown its hat into the IP PBX ring with the release of its GXE502X SIP PBX Appliance. The GXE502X comes in a 4FXO and 8FXO version, allowing easy integration of analog (POTS) PSTN trunks. The product also carries a price point which should make it appealing to small businesses with a limited budget. All the typical IP PBX features are present, including autoattendant, voicemail, configurable dial plan and one-touch provisioning SIP trunking support.

Grandstream port

Grandstream is also readying a 24 port FXS SIP gateway, the GXW4024….offering 24 analog connections for legacy PBX and analog endpoint integration with Asterisk and other SIP platforms.

Polycom

Polycom has introduced two new SIP conference phones, both supporting their new HDVoice wideband audio codec.

Polycom SIP Conference Phone

The Soundpoint IP6000 figures to replace the popular IP4000 model. The Soundpoint IP7000 incorporates a sleek new form factor, and a few features not found in the 4000/6000. Both conference phones also offer native support for 802.3af Power over Ethernet, which is an appreciated addition.

Digium

Digium

Digium has several new product additions. First off, they have released a new, small form factor solid state appliance to host their popular Switchvox SOHO and SMB Edition IP PBX software. The new AA60 supports Digium FXS/FXO TDM cards, and supports up to 10 concurrent calls. The AA60 is a directly replacement for the Switchvox midtower server, which is no longer available.

Digium PCI FXS FXO cards

Digium has also replaced their popular, low-density TDM400 series PCI FXS/FXO cards with the new TDM410 series. The form factor is very similar, but the cards are now available with optional Octasic hardware echo cancellation module.

Aastra

Aastra

Canadian firm Aastra is also readying an IP PBX appliance solution based upon Asterisk Open Source software. The Aastralink Pro 160 offers a wide range of PBX functions, supports SIP trunking, and offers up to six integrated FXO ports for PSTN Analog POTS integration.

VoIP Supply Wins ‘Best Place to Work’

April 9, 2008 by Ben Sayers

Best Places to Work
Yesterday VoIP Supply was awarded the Best Place to Work in Western New York (for companies with 51-100 employees).

This is a tremendous honor and a testament to the quality of staff employed at VoIP Supply. Growing at such a rapid pace, constantly adding more and more people and always having to change in order to set the pace and stay on top of our game is a challenge. Ensuring that the workplace is safe, fun, and enjoyable while maintaining a high level of morale makes that challenge that much more exciting.

So far this year we have added 23 new people (more than 50% growth), changed our ERP and CRM platforms, expanded into additional office space, refined and documented all company processes and procedures, and expanded into two additional markets (IP Surveillance and Voice Recordings).

I am proud to be part of an organization that can pull off all of these things, do it with a smile, and look forward to coming back in tomorrow. To all those at VoIP Supply, thank you for being part of this team.

The Exit Interview and Beyond

April 7, 2008 by Ben Sayers

Always an interesting component of the CEO’s role is the summarization of one’s employment at the time of departure. These are usually quite different when the employee is leaving on their own versus when they have been terminated. We conduct exit interviews whenever possible to determine the reasons for the change from the employee perspective and to learn how we can improve if necessary.

“The grass is greener.” Often times people leave the company because they feel that there is a more suitable opportunity elsewhere. In many companies I am sure this is true; at VoIP Supply the owners and managers work very hard to create an environment where people can grow their own green grass and not feel a need to leave. If the greener grass is more money for less work, responsibility and accountability then there will always be a short term gig with skeletons in the closet. This is not a new concept and sometimes there may well be greener grass depending on where you are coming from. Many have left VoIP Supply to pursue greener pastures. Most have come back to seek their old spot or continue their quest, also embarrassed to return once they found that the grass was green but spoiled.

“I was not treated fairly.” What is the benchmark for fairness? It is pretty self explanatory yet there are different measurements based on experience, responsibility, competence, expectations, seniority and level of compensation. Being treated fairly, at least at VoIP Supply, is generally an issue of perception. Not liking the way you were treated is different than not being treated fairly. Times change, people change and the needs of the business change; as a result it can often be seen as unfair when something that was acceptable or tolerated 12 months ago is no longer going to cut it. When the needs, attitude, and direction of the company are in direct opposition of your own needs, attitude and agenda; that is when it becomes easy to take a stance of not being treated fairly. It takes a lot of time and energy to fairly and consistently manage 65 people, and to work to ensure management is on the same page when it comes to fairness and consistency. In short, fairness is often based on perception and not reality, and it is a two-way street where fairness must go both ways – in the reality wagon.

“Too much was asked of me.” America has built an economy where too much is asked of everyone, work weeks are no longer 40 hours long and the expectation is that everyone is to perform at a comparable level, even after a superstar comes in and raises the bar. VoIP Supply is not too different, as a lot is asked and expected of everyone; the difference lies in understanding an individual’s capabilities versus attitude and drive. When someone quits because “too much was asked of them,” it is always a sign of a poor attitude, lack of motivation or unwillingness to be part of a fast-paced, rapidly growing, constantly evolving environment. You know what? That’s ok; the world needs those people too, just not here at VoIP Supply. Those driven to succeed, grow and be successful are held in extremely high regard.

Polycom Releases New Software and Updates

April 4, 2008 by Garrett Smith

Polycom rolls out software updates

We received some information from Polycom this week about some software updates and a new productivity application suite. These can definitely help all you Polycom lovers out there, and VoIP Insider is happy to spread the word.

Available Polycom software updates

An updated software release, SIP 3.0.1, is available from the Polycom Resource Center. This is a maintenance patch release and includes improvements related to issues that have been uncovered in the field. The Release Notes are available from the Polycom Support website.

An updated software release, SIP 2.1.3, is also available from the Polycom Resource Center. This is also a maintenance patch release and also includes improvements related to issues that have been uncovered in the field. This release is intended for customers that have SoundPoint IP 300 or 500 VoIP phones deployed as these are not supported on the SIP 2.2.x or SIP 3.0.x releases. The Release Notes are available from the Polycom Support website.

Polycom Productivity Suite

The Polycom Productivity Suite will be shipping around April 2. The Productivity Suite is a set of five productivity applications designed to enhance a user’s overall productivity and improve their user experience when using Polycom’s SoundPoint IP and SoundStation IP phones. The applications include:

Corporate Directory Access, which eliminates directory duplication through integration of the phone with any LDAP compliant corporate directory system (this includes support for Microsoft’s Active Directory)

Local Call Recording, which allows any active calls to be instantly recorded using any standard USB flash drive

Visual Conference Management, allowing a user to locally conference in 3 additional participants into a 4-way local conference call, which can be completely managed via an intuitive visual menu

Voice Quality Monitoring, providing the IT community with a way monitor call quality in real time and proactively detect voice quality monitoring issues

Third Party Call Control, allowing a third party application to control the phone as well as to obtain presence information from the phone

More information on the Productivity Suite can be found at Applications for SoundPoint IP Phones page at the Polycom website.

How Are You Facilitating Your Inbound and Outbound DID’s?

April 3, 2008 by Garrett Smith

Direct Inward Dialing and how it works

Many of us may have questions about how DIDs work and how to provision them. DID stands for “Direct Inward Dialing”. DIDs are typically used in conjunction with an IP PBX, to route incoming and outgoing calls to their correct source or destination. Almost every IP PBX has a method of facilitating DID’s, whether that be internal or external to the server. However, products used to facilitate them are often in question.

There are three ways that most businesses are “bringing in” their DID’s. The first method is Analog Trunking or “POTS” (Plain Old Telephone Service). Analog Trunks may be comprised of physical copper PSTN lines paid for and supplied by your local telephone company. These are pure RJ-11 analog lines, no different from your “landline” wall jack at home. Most SOHO applications utilize analog POT’s lines since they are more cost effective. The typical number of physical PSTN lines is usually around four to eight, and will vary depending upon the number of inbound/outbound calls the business needs to support. Each physical POTS line is equal to one channel, and represents a 1:1 ratio. Each physical PSTN line also has a single DID number associated with it. If you have four analog POTS lines, you have four DID’s or channels available to make inbound and outbound calls.

Removing Voip from the picture for a second.

Take VOIP out of the picture for a second…. in a true analog environment, each PSTN line would be connected to an analog telephone (user), and that telephone would be associated with a specific DID number. Let’s bring back VOIP now….analog telephone lines are NOT connected to the phones themselves, since in most cases, you will be using VOIP phones, but rather into the central location… the IP phone system. Analog lines are facilitated within the IP PBX via FXO PCI cards. If you have eight incoming RJ-11 PSTN lines, you will essentially need an eight-port RJ-11 analog PCI card much like the Sangoma A20004D or Digium AEX808E. Simply connect each RJ-11 connection into the ports on these cards, most IP PBX’s will auto-detect their presence, and you are now permitted to configure your analog trunks or channels within the IP PBX. Since the analog DID’s are now facilitated at a central level, they are not on a pure 1:1 basis because when a VOIP phone accesses this trunk group to make an outbound call, it is not associated to that one DID always, it will simply grab the next available DID within the channel group and use that. This allows users to add more VOIP phones to the scenario without physically increasing their number of POTS lines. Essentially, if you have 16 VOIP users, but only experience around 8 concurrent calls at a time, you would only need 8 POTS lines, rather than 16. Please note, you are not limited to four or eight-port analog PCI cards. These numbers were offered as a very basic example. Please check out Sangoma and Digium on voipsupply.com for further clarification on these cards.

Digital connections are becoming very popular amongst larger organizations because of ease of use and cost savings over Analog POTS Trunking. In most large applications, there is a need for 24, 48, or even 96 + voice channels. The easiest method to facilitate this number of voice channels is via digital T1 lines using a T1 provider. Most IP PBXs have the ability to integrate digital T1 connections, either through T1 digital PCI cards or external T1 gateways. A T1 is an essential 24 individual lines (equivalent to 24 analog POTS lines) delivered over a single pipe. A T1 is configured at the IP PBX level through digital trunk groups or channel groups.

2 Types of T1 Provisioning

T1’s can be provisioned in two flavors. The first is through PRI signaling (Primary Rate Interface). PRI’s contain 24 channels in total but only allow for 23 configurable channels. The remaining channel is a “work-horse” channel so to speak, performing all of the overhead signaling work for each of the 23 available channels. The second method of T1 signaling is a method called CAS (Channel Associated Signaling). CAS signaling with the “T” allows for 24 channels to be configured as voice, data, or both. Each channel performs its own work to allow for proper signaling to take place. Please check with your T1 provider to ensure which method they are using, and opt for the best method to fit your needs. Digital T1 lines are integrated with an IP PBX very easily….simply connect the T1 to a Sangoma A101 card or Digium TE-122P card, which are both Single T1 cards. However, you are not limited to a single T1; a standard digital PCI card can have up to four T1 ports incorporating 96 channels within the system, but that doesn’t mean you can’t add a second quad T1 card. For those larger T1 applications I just spoke about, please refer to the Sangoma A104D card or the Digium TE-420B.

SIP Trunking

The final method to bring in your DID connections is through a method called SIP Trunking. SIP Trunking is done completely over a data connection (Typically T1, Fiber, ADSL or Cable), and is provisioned to your PBX through a connection to the WAN and your SIP Trunk provider. SIP Trunking is becoming one of the most cost-effective methods of acquiring inbound and outbound channels because there is no need for a physical connection on the premises. The only down-side? Certain providers will only provide local SIP Trunks to specific geographic locations. Check with your prospective VoIP provider for availability of SIP Trunking services in your area. SIP Trunking is quickly gaining in popularity with businesses of all sizes. I think you will see more providers start to offer this solution, if they don’t already, and we should see SIP Trunking to almost every local DID as time and demand progress. Everything nowadays seems to be “up in the clouds.” SIP Trunking is no different, and is conforming to the future specifications of what consumers, both large and small, are expecting.

New Redfone Communications Product Alert

April 2, 2008 by Garrett Smith

Redfone Communications New Releases

We have added two new Redfone Communications products to our ever expanding Voip Supply Catalog. The first is the foneBridge2-EC Single which is a T1/E1 PRI-to-Ethernet Bridge. It is an integrated black box “appliance” designed to streamline installation and enable redundant design of Asterisk based VoIP/PBX systems. The second is also a RedFone Communications offering. It is a foneBridge2 Single T1/E1 Bridge Asterisk PBX T1/E1 Redundancy, High Availability & Load Balancing in an economical Enterprise class solution. As always you can either purchase online or call one of our friendly sales engineers with any further questions you may have.

VoIP Supply Unleashes Go3 Guaranteed Replacement Warranty Program

Go3 allows customers an affordable way to protect their hardware investment

Go3 Guaranteed Replacement Warranty

VoIP Supply announced the launch of its new three-year, no-questions-asked, guaranteed replacement warranty, Go3. The new Go3 Guaranteed Replacement Warranty gives customers a no hassle way to protect their VoIP hardware purchases. The new warranty program is based off of VoIP Supply’s existing warranty program and was redesigned with feedback from existing customers.

“Our customer service is VoIP Supply’s number one priority,” said Benjamin P. Sayers, CEO of VoIP Supply. “We have been successful due to our wonderful customers’ trust and support. The Go3 Guaranteed Replacement Warranty is just another way that we can bring our clients needs to the forefront.”

The Go3 Guaranteed Replacement Warranty allows customers to protect their VoIP Supply hardware investments. Their hardware is completely covered for a full three years from the date of purchase. If the equipment fails, malfunctions, or ceases to operate properly for any reason (manufacturer defect, component failure or user negligence), VoIP Supply will provide a replacement.

Unlike most extended warranty programs offered by retailers, with Go3 there is no fine print or run-around during the redemption process. The only conditions include purchasing the equipment from VoIP Supply and returning it within the three-year period. The coverage must also be purchased at the point of sale, and it is not refundable.

For more information about the VoIP Supply Go3 Guaranteed Replacement Warranty, please visit the go3 warranty for more information.