The Jabra Solution for Knowledge Workers in Noisy Offices

November 7, 2014 by Nathan Miloszewski

The downside to producitivity is noise.

If your office is overflowing with busyness and you could benefit from some (relative) peace and quiet in your cramped workspace, then the Jabra EVOLVE series of headsets might be the answer.

Check out this article from Jason Perlow (@jperlow) at ZDNet who provides a full review of these new Jabra Evolve 80 noise cancelling Unified Communications (UC) headsets:

Jabra’s Evolve 80 is not by any means a budget VOIP headset that you want to issue to an entire fleet of office workers. But it is definitely a product you want to treat yourself to, particularly if you do heavy conference calling at your desk, own mobile devices, and need to be able to “tune out” when necessary.

If you think that this over-the-head style is bulky, take a look at this other UC headset from Jabra; the Jabra Motion UC Bluetooth.

The Jabra Motion is a sleek monaural headset that lets you work hands free from all of your devices (cell, tablet, laptop, softphone). Click here to see the full Jabra Motion series with models also designed for Microsoft Lync.

Via ZDNet

New Digium Digital Cards Feature PCI Versions with TE236 and TE436

November 6, 2014 by Nathan Miloszewski
Digium Digital Cards

New Digium TE236 and TE436 Digital Telephony Cards offer PCI Connectivity

Digium has expanded on their previous TE23x series with the new addition of TE236 and TE436 digital telephony cards.

The most noticeable change is that the TE236/TE436 are PCI versions of the previous PCI Express TE235 and TE435 models.

These new cards still have the same low profile and half-length design that you’re used to and are available in both dual span and quad span formats with optional hardware echo cancellation.

The echo cancellation cards are denoted by the “BF” at the end of the model number (TE236BF, for example).

Dual Span Digium TE236 Features:

  • 2 x RJ48 Interface Ports
  • Up to 48 (T1/J1) or 60 (E1) simultaneous calls
  • Selectable T1, E1, or J1 Mode
  • PCI 3.3V/5.0V
  • Hardware Echo Cancellation with TE236BF model

Quad Span Digium TE436 Features:

  • 4 x RJ48 Interface Port
  • Up to 96 (T1/J1) or 120 (E1) simultaneous calls
  • Selectable T1, E1 or J1 Mode
  • PCI 3.3V/5.0V
  • Hardware Echo Cancellation with TE436BF model

Who Are These For?

The TE236 and TE436 cards are designed for:

  • Enterprise level phone systems,
  • VoIP conversion from legacy PBX,
  • IVR trees,
  • Meet-Me Conference bridges,
  • VoIP Gateways
  • Calling card platforms

Warranty and Satisfaction Guarantee

Digium products are backed by a 5-year warranty and the risk-free Digium Exceptional Satisfaction Program (ESP):

Under the Exceptional Satisfaction Program (ESP) risk-free, Digium Quality Guarantee, Digium® will refund the purchase price of any qualifying telephony card, Switchvox appliance, Asterisk Add-on, Digium Phone, Gateway, or R-Series to any customers who are not 100% satisfied that the Digium product(s) they purchased performed as designed.

For more information call us today at 800-398-8647 or reach out to one of our VoIP Experts.

Digital Disruption Conference Helps Businesses in Digital Age

October 30, 2014 by Nathan Miloszewski

Kodak, founded in the late 1800s, was the company that made it easy for anyone to snap a picture with a camera.

Over 100 years later the company had to declare bankruptcy. The Kodak name, synonymous with photography and film, is now also known for being slow to adapt to digital technology.

That’s the unfortunate side of digital disruption and the subject of a new book called iDisrupted by John Straw and Michael Baxter.

Jerin Mathew writes about the book in this IBTimes article, Digital Disruption to Remove Many Large Companies from Top 100 List, where Baxter explains:

“In our book, we try to explain why it is that technology is set to change the world like it has never been changed before. This is exciting, but it is also scary. There will be winners and losers, and some of the world’s largest companies will be amongst the losers.”

TM Forum’s Digital Disruption Conference

What happened to companies like Kodak is one side of digital disruption.

With change though there’s an opportunity to take a look at your business model and ask your company what it can do to stay relevant in your marketplace.

One resource you can take a look for ideas is TM Forum, a trade association that helps enterprises, service providers, telecom, software suppliers, and others “transform to succeed in the digital economy.”

TM Forum is holding their Digital Disruption 2014 conference from December 8-11, 2014 in San Jose, California which focuses on:

…the business and operational issues of delivering digital services in today’s disruptive market. While most conferences focus on the underpinning technologies, Digital Disruption looks at the business issues of how advanced IT can help players to innovate more effectively; adapt more rapidly; deliver more for less and exploit major new opportunities.

There’s a wide range of speakers at the event from companies like Nordstrom, Philips, Microsoft, Stuff Magazine, AT&T, Symantec, Huawei, and lots more.

The key topic of Digital Disruption 2014 is innovation, specifically:

  • Disruptive Innovation:  Addressing the opportunity and the business model challenge, changing the customer experience
  • Tackling Innovation: Man versus machine, employee and organizational culture change
  • Monetizing Innovation:  Monetizing & humanizing Internet of Things, driving engagement and delivering results

First Look: Grandstream UCM6510 IP PBX Appliance

October 29, 2014 by Jeff Quinn

Grandstream UCM6510

Introducing the new Grandstream UCM6510 IP PBX.

What is it?

The Grandstream UCM6510 is a new IP PBX appliance with enterprise grade features that is still targeted for the SMB market.

This IP PBX appliance is similar to Grandstream’s UCM6100 series, but with a much higher user count. The Grandstream UCM6510 is the most robust and feature appliance that Grandstream has released.

The UCM6510 offers a single T1/E1/J1 along with 2 FXO and 2 FXS ports in case of a power outage. Grandstream made this IP PBX appliance come standard with dual gigabit Ethernet ports, a 1 GHz quad-core Cortex A9 application processor along with a customizable auto attendant.

Who is it for?

The UCM6510 has call center and enterprise level features and can also be scaled down and used in a small-to-medium (SMB) setting.

With that in mind, plus the ease of use and set up that Grandstream provides with their appliances makes these units attractive to a large array of businesses and users.

With call features too long list and a price point that one would think it’s a mistake, the Grandstream UCM6510 is a great product for most any type or size of business.

Pricing

Grandstream UCM6510 IP PBX Appliance – MSRP $1,999

Grandstream UCM6510 IP PBX Appliance Features:

  • 2 FXS and 2 FXO ports with lifeline capability incase of a power outage
  • Single T1/E1/J1 port
  • Ability to handle unlimited SIP accounts
  • Rack mount available
  • Call Center features accessible
  • Up to 2000 registered SIP endpoints and 200 concurrent calls
  • Auto Provision and Detection of end points
  • Redundant power supply

Availability

You can shop on-line for Grandstream on VoIPSupply.com or contact the VoIP Supply sales team at 800.398.8647

Tune Up Your Sangoma TDM Cards with this Online Training

October 24, 2014 by Nathan Miloszewski
Sangoma Training

Online Training: Get Superior Quality and Control – Performance Tune Your System

Sangoma has been offering free online technical training courses to introduce you to their line of Sangoma TDM interface cards.

The first session was a run-down of available options and how to choose the right telephony card while the second session dealt with how to install Asterisk with Sangoma.

Sangoma Training Session #3

The next, and final, training in this series is called, Get Superior Quality and Control – Performance Tune Your System.

  • System Design Considerations
  • Advanced Troubleshooting Techniques
  • Physical and Signaling Layer Issues
  • Q&A Period
Sangoma Training

Click to Register

Block out 90 minutes for this training and come prepared with your Sangoma questions as there will be a Q&A period at the end.

How, Where, When to Register

Click here to register for Session 3 of 3:

  • Performance Tune Your System

This training will be held on:

  • Day:  Wednesday, October 29, 2014
  • Time:  8 AM – 10 AM EST

What’s New at Astricon 2014

October 23, 2014 by Nathan Miloszewski

AstriCon 2014 is happening right now in Las Vegas from October 22nd to the 24th, so if you’re not there here’s why you should attend next year.

Over the course of three days the annual AstriCon conference aims to fulfill its mission of expanding Asterisk expertise through these five areas:

  • People
  • Presentations
  • Panels
  • Products
  • Projects

For fans, developeers, and businesses – this is the place for anyone who uses the Asterisk telephony platform to learn about the latest projects, updates, and technical details.

Hackathon

The first ever AstriCon Hackathon was held this year for communications designers and developers to challenge themselves in a fast-paced app designing contest with custom software built on Asterisk, Respoke, and Clarify.

Here’s a recent video that explains more about it:

What’s Respoke?

Also new at AstriCon this year is Respoke which is Digium’s new cloud-based platform that gives developers another tool for the growing use of web communications.

The Respoke API wants to make it easy to add voice, video, messaging, and data to applications.

Introducing CloudSpan by VoIP Supply SIP Trunks, Hosted PBX, and Hosted Fax

October 17, 2014 by Nathan Miloszewski

For over 12 years now you’ve trusted us for all the phones, adapters, switches, and gateways that you need for your VoIP network.

We want to help you even more.

So we built our very own VoIP services offering from the ground up with future proof, scalable, and flexible features that we know you need from your communication platform.

Introducing CloudSpan by VoIP Supply – our first ever cloud-based VoIP service that include:

  • CloudSpan SIP Trunks
  • CloudSpan Hosted IP PBX
  • CloudSpan Hosted Fax

About CloudSpan SIP Trunks

If you manage your own VoIP system, CloudSpan SIP Trunking offers business class SIP trunks with standard features such as unlimited inbound calling, e911 Emergency Service, caller ID with location, HD voice, and fraud protection.

CloudSpan SIP Trunking service does not require a contract and there are no setup fees and no termination fees.

About CloudSpan Hosted IP PBX

For businesses that don’t want to worry about managing a communications platform, CloudSpan Hosted IP PBX professionally maintains the entire phone system in the cloud, removing the customer’s burden of purchasing and maintaining IP PBX hardware and software.

CloudSpan Hosted PBX has the flexibility to scale up or down as your calling needs change with features geared towards today’s mobile workforce including unlimited extensions, cloud extensions, find-me / follow-me call routing, e911 Emergency Service, call recording, voicemail to email, remote and mobile access, and 24/7 system monitoring and management.

About CloudSpan Hosted Fax

Like it or not, you probably still have to send and receive faxes to and from your customers and clients.

CloudSpan Hosted Fax simplifies this paper-based technology into a digital, cloud-based solution by allowing you to send an receive faxes via email, web broswer, or even a smartphone app.

Save money by getting rid of your dedicated phone line for faxing and never worry about fax machine maintenance again.

Brand Promise

Love is free. So is your service if we let you down.

– Ben Sayers, VoIP Supply CEO

We want you to be completely satisfied with CloudSpan.

If we let you down, we’ll refund your last payment and help you switch to a new service provider.

Plus there are never any contracts, setup fees, or termination fees.

Need More CloudSpan Info?

Whether you’re new to cloud services or just need more information about CloudSpan, check out these links:

What’s the Risk Using Asterisk: Is this Open Source VoIP Platform Safe from Hackers?

October 16, 2014 by Nathan Miloszewski

Asterisk VoIP Security

When I came across a blog on Huffington Post that called Asterisk out on the security of their open source VoIP platform I just had to know, is this true?

So I asked Asterisk (after I said “asked Asterisk” five times fast) and got this detailed response from David Duffet, Director of Worldwide Asterisk Community.

Duffett (@dduffett) explains that protecting your network is a not whole lot unlike fortifying your house against break-ins.

VoIP Supply: Who is the Asterisk VoIP platform designed for?

David Duffet: The Asterisk IP communications engine is for anyone that wants to create a flexible and powerful communications solution. Asterisk configuration is performed through a number of ascii text files, and this is why a number of pre-packaged IP PBX solutions based on Asterisk have become available that allow configuration via a web GUI.

VS: Why open source?

DD: When Mark Spencer (the creator of Asterisk and CTO of Digium) decided to make Asterisk an open source project, he did this in part to liberate the stodgy, closed world of telecoms, but also to allow (and encourage) contributions to Asterisk from people all over the world that are particularly keen to see Asterisk enhanced in specific directions (like conferencing and contact centre applications).

VS: In this blog post on Huffington Post, 6 Keys to a Successful VoIP Implementation, the writer, Jason Volmut (@javolmut), CEO of CPUrx, states that:

“VoIP systems built on the open-source telephone platform Asterisk are routinely subject to hacking attempts, and should be avoided. “

What VoIP security measures can Asterisk take to secure their systems from hackers?

DD: Although there are a number of places within Asterisk that could be configured to enhance security, I would like to make some more general points:

The mention of only Asterisk in point 5, regarding security, is extremely misleading.
To set the scene, PBXs, even before the advent of IP communications, have always been subject to attacks of one sort or another – all the way from people trying to hack into voicemail boxes to full scale toll fraud through PRIs or even analog lines.

*ANY* SIP IP PBX that has an open connection to the internet (i.e., not within a VPN, or not tied down to a specific IP address, or addresses) will be subject to hacking attempts.

" Just like any type of system – it’s all in the implementation. If that is done in a sloppy way, it could lead to trouble." - David Duffett, Asterisk

” Just like any type of system – it’s all in the implementation. If that is done in a sloppy way, it could lead to trouble.”
– David Duffett, Asterisk

Asterisk is certainly the most popular and established open source communications engine in the world, with millions of Asterisk-based IP PBXs out there – but they are by no means particularly prone to issues of this nature. Just like any type of system – it’s all in the implementation. If that is done in a sloppy way, it could lead to trouble.

There is lots of information around on the internet about certain brands of proprietary IP PBXs and potential vulnerabilities, but to focus on the PBX is to miss the main point about securing IP systems – and that is to ensure proper measures are taken at the network level, before thinking of applications running in the network like a PBX or a CRM system.

If you found a robber in your kitchen, you know that he would have broken into your house through the front door, back door or a window. The best thing to do would be to improve the security on the exterior of your house so as not to let the robber in! And so it is with your network… Stop the bad guys getting into your network in the first place!

Anything you can do in a given appliance or application like an IP PBX or a CRM system should be seen as a secondary line of defence.

Due to the power and flexibility of Asterisk, there are actually more things you can do on an Asterisk PBX to detect and prevent any form of compromise than there are on any other PBX solution. Of course, they must be implemented and adjusted by people that know what they are doing.

How Working From Home Caused Michael Graves to Start Writing about VoIP: “I had to come up with my own IT strategies, including telephony.”

October 10, 2014 by Nathan Miloszewski
Michael Graves & Shdow-IMG_5786

Michael Graves, his dog, and 100+ lb. pumpkin. Photo Courtesy: Michael Graves (www.mgraves.org)

Office technology is supposed to make our working lives easier but the overwhelming amount of options makes it difficult just choosing the right device.

So where do you turn and who do you trust when you want to know who’s used this stuff before, what works and what doesn’t, and how much hair will I pull out trying to install this thing?

This is when you seek the guidance of a VoIP blogger like Michael Graves (@mjgraves) who runs the site Graves on SOHO Technology. He tests products geared for the small office / home office (SOHO) user and offers you, the reader, a wealth of detailed product reviews, guides and how-to articles, and great hands-on advice.

Graves explained to us that products under review spend a considerable amount of time in use on his desk before he even thinks about writing about them, so you know you’re getting quality insight. Which is great, but we wanted to know, “Why?” So we asked him a few questions about how it came to be that he created his site with all these great resources and what he thinks about the future of VoIP.

VoIP Supply: Tell us a little bit about who you are and what Graves on SOHO Technology aims to provide for your audience.

Michael Graves: My professional background includes working as a video editor and graphic designer. I also spent a long while on the technology end of the broadcast business for an English company that builds specialty graphics equipment. That work, mostly system integration and technical support, gave me the opportunity to travel extensively, which was great exposure to a variety of corporate IT infrastructures and strategies.

Early in my career I took a bit of time away from television production to write for a magazine. That exercise gave me the confidence that I could go beyond design and layout to actually write content. It was much later that I started the blog as a way to share the experience gained over years of working from my home office.

VS: How did you get started in VoIP?

Michael Graves

“I was one of the earliest people clamoring for an embedded Asterisk appliance. That’s what started me writing about VoIP. “

MG: Working as I did for a UK based firm, with limited US presence, I was based in my home office in Houston. I wasn’t just occasionally working from home as is more commonplace. I was and remain, 100% home office based. That means that I had to come up with my own IT strategies, including telephony. We had four analog phone lines, and watched as the cost went up while their capabilities were flat-lined.

Eventually, I thought that I needed to be able to do more with the funds. That drove me to look at Asterisk in it’s pre-v1.0 days. After some experimentation, I used a small Asterisk installation with some Polycom and snom phones to replace the small Panasonic KSU that was providing our home and office phones.

I was one of the earliest people clamoring for an embedded Asterisk appliance. That’s what started me writing about VoIP. I documented the process of building an Embedded Asterisk server using a Soekris single board PC and the then fledgling Astlinux embedded distribution. I wrote that for Tom’s Hardware (which later became Small Net Builder) late in 2005.

Since I had analog lines I needed FXO interfaces. My experience with early FXO cards was pretty horrible. Even the new Sipura SPA-3000 (I was in the beta program) was less than wonderful. I eventually ported my numbers to an ITSP, putting all of our voice calling over a DSL circuit.

Back then DSL was pretty slow. It took patience to select and tune a router to ensure good voice quality over DSL. The whole process or moving to Asterisk using voice-over-DSL is what drove me to start the blog. I was answering questions in various forums. It seemed like the same handful of questions kept being asked all the time, so I thought I’d carefully write down my experience in a place that I could use as a reference.

VS: What are your favorite things about VoIP, or some of the creative ways you’ve seen people use the technology?

MG: I think it’s wonderful that we can take standards compliant hardware and software to build solutions to real problems. Sometimes it’s driven by saving money, but for me it’s more often driven by the desire to solve a problem in a new and interesting manner.

People have done all kinds of fun and interesting things with VoIP tools. My former employer had to exhibit at a major trade show in Las Vegas every spring. They paid handsomely for the privilege of internet access at the booth in the exhibit hall. Then they paid again to have a voice line dropped into the booth. Then they paid a third time for calling to the UK head office where their team of engineers were busily trying to finish the code that was to be used for the show. It was silly, bit it was the way of the world.

Once they were familiar with what I had done in my home office they asked me to bring a small PC and a couple of SIP phones to the show. We never again had the convention center install the voice circuit. Better yet, we could have multiple simultaneous calls ongoing over the SIP service, including calls from staff in the UK who had a few of their own snom hard phones by then. We had better calls using G.722 based wideband audio, and we paid less. That’s win-win.

Looking to the present-day and forward, I really like that both HDVoice and video are coming into more widespread use. It’s been about to happen for many years, but it’s really picking up momentum now. Even the US mobile carriers are now supporting HDVoice in limited ways.

Graves answers and shares a readers question.

Graves answers and shares a readers question.

VS: What trends are you seeing now and any predictions for the future?

MG: I expect to see growing use of hosted PBXs by small businesses. That’s what I use myself. There’s a chance that Microsoft could step into this space with a new hosted Lync offer.

I don’t think that the desk phone is going away just yet, at least not in the world that I inhabit. I still need to be on the phone even as my desktop reboots.

More use of video. We see it already with Hangouts and various other services. WebRTC has yet to really make a dent into anyone’s business, but it’s a tidal wave that’s going shake things up a lot.

On the other hand, I’m not that enthusiastic about the more costly video conferencing end-points. I think that their days may be numbered. Low-cost end-point psuedo-appliances like Chromebox for Meetings may do well, but it’s too early to tell as yet.

I hope that we see more use of encryption in all aspects of telecom. WebRTC may be a big step forward in that direction.

VS: Writers read a lot. What’s on your reading list; websites you love or books you’d highly recommend?

MG: It was pain making the switch from Google Reader when it was shuttered. These days I meet my morning using Feedly to catch up on the news of the day. While I have a couple hundred feeds collected there, I’d recommend Light Reading, No Jitter and UC Strategies for telecom topics. Also, Dave Michels who also writes at www.talkingpointz.com is someone I really admire. Avaya’s Andrew Prokop has some great insights into technical matters at SIP Adventures.

As for books, right now I’m trying to stay focussed on the Certified Specialist of Wine Study Guide. I’ve just completed a class and I’d like to take the test some time in the next month or two. It’s a grueling test, requiring 75% correct to pass. I’m told that only one-in-three people pass on their first attempt. Everyone needs a hobby, right?

There are two books I read that I’d recommend to anyone; “Exploding the Phone: The Untold Story of the Teenagers and Outlaws who Hacked Ma Bell” by Phil Lapsely and “Perfecting Sound Forever: An Aural History of Recorded Music” by Greg Milner. Both are excellent accounts of the lengthy history of something that most people take for granted.

3 Technical Training Sessions for Sangoma TDM Interface Cards

October 7, 2014 by Nathan Miloszewski
Sangoma A100 Cards

Sangoma TDM Card Training Sessions Now Available Online

Sangoma, a leading provider of voice and data cards since 1984 for VoIP platforms like Asterisk and FreeSWITCH, is now offering free online technical training courses to introduce you to their line of Sangoma TDM interface cards.

What You’ll Learn at Sangoma TDM Training

Sangoma offers a wide range of telephony cards including digital voice cards, analog voice cards, wireless GSM cards, hybrid voice boards, and ISDN BRI voice cards.

The first of three training sessions will help you learn:

  • Which Sangoma card you should choose (Analog, BRI, PRI, E1, T1, J1, GSM) and the benefits for each type in your system.
  • Introduction to Sangoma’s WANPIPE Driver including:
    • Installation methods for Asterisk based systems
    • How to configure your Asterisk system with Sangoma
    • Hardware installation details

Block out 90 minutes for this training and come prepared with your Sangoma questions as there will be a Q&A period at the end.

How, Where, When to Register

Click here to register for Session 1 of 3:

  • Making the Right Choice: Introduction to WANPIPE

This training will be held on:

  • Day:  Wednesday, October 15, 2014
  • Time:  8 AM – 10 AM EST

Next Up

Here are the next Sangoma training sessions in this series:

  • Part 2: Installation with Asterisk | Tuesday, Oct. 21, 2014 | 8 AM EST
  • Part 3: Performance Tune Your System |  Wednesday., Oct. 29, 2014 | 8 AM EST
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