VoIP Supply Pitches In for 22nd Annual United Way Day of Caring in Buffalo and Western New York

September 20, 2014 by Nathan Miloszewski

day of caring_voip_supply_2014_3

This past week on September 17th and 18th, VoIP Supply participated in the 22nd Annual United Way of Caring in what has become one of the biggest community service events in Buffalo and Western New York.

It’s a great event, with even greater turnout, where companies lend out their employees who get a chance to help their neighbors and gain a some team-building experience.

We were proud to help out Opportunities Unlimited of Niagara with blacktop sealing and another project where a resident was in need of fence staining.

Get Involved, Year of Giving

Visit the United Way of Buffalo on Facebook or their website for more information on their charitable efforts and ways that you can get involved.

We look forward to next year’s Day of Caring but in the meantime, VoIP Supply’s Year of Giving continues.

 

How Does Net Neutrality Affect VoIP?

September 18, 2014 by Nathan Miloszewski

If you’ve been following the net neutrality debate you know that the FCC is currently reviewing over 3 million comments from the public to find out how they feel about the current Open Internet standard that we’re used to.

So far, public opinion is less than 1% against net neutrality. Most people, it seems, believe that all internet traffic was created equal.

Can the voice of the people be enough to sway the FCC or will service providers be allowed to monetize priority internet traffic?

We’ll have to wait. In the meantime, if you’ve wondered how this will affect your Voice over IP, here’s a helpful infographic from Visually:

(more…)

VoIP Q&A: Failover Port on Cisco SPA232D ATA

September 16, 2014 by Nathan Miloszewski

Questions about VoIP devices and services are regularly submitted to VoIPSupply.com through a technical support ticket or via the “Ask The Expert” tab on our product pages.

We respond to these requests directly but more often than not, this Q & A would be helpful for lots of other folks.

Below are your VoIP questions answered – Real questions from real people just like you.

Q: In case that my Internet Provider or SIP provider fail, has the Cisco SPA232D the survival option through it’s FXO port?

Cisco SPA232D

Cisco SPA232D Analog Telephone Adapter with Built-In DECT

A:   The Cisco SPA232D ATA is an FXS/FXO gateway with DECT so that in addition to using your analog phones and fax machines with VoIP you can, with the addition of an integrated DECT base station, also add up to five Cisco SPA302D handsets to communicate with it.

In regards to your question, the answer is simply “Yes.”

For more detail I will refer to the post of one of our former colleagues who wrote the post, FXS and FXO – You should be in the know:

Some ATA’s also have an FXO port. This connects to the wall jack by the computer to provide failover. In this case, in the event of internet failure, you can still make telephone calls via the POTS line. Also, you can use the FXO port to make free local calls on your POTS line.

FXO ports are most often used on Gateways and PBX’s to support Failover or Fallback as mentioned above.

For more information on all of the FXO control settings, check out the Cisco SPA232D Datasheet.

Cisco SPA232D ATA shown with SPA302D DECT Handset

Cisco SPA232D ATA shown with SPA302D DECT Handset

 

Stay Tuned

Check back next time for more VoIP Q & A.

Thanks for your questions!

Xorcom Interviews Explain Benefits of New CompletePBX Version 4.5 and Orion Video Conferencing for SMBs

September 11, 2014 by Nathan Miloszewski

IT Expo West 2014 is over but Xorcom has shared a couple great recap interviews that explain two of their solutions:

  • New, soon-to-be-released Xorcom CompletePBX Version 4.5 that will offer stronger security features and an open source endpoint manager
  • Xorcom Orion Video Conferencing for SMBs that’s affordable and reliable

Xorcom Orion Video Conferencing for Small Businesses

Via Xorcom

Shipping is On Us for Digium Digital Cards

August 26, 2014 by Nathan Miloszewski

Digium Free Shipping

Right now VoIP Supply is offering Free FedEx 2nd Day Shipping* on our most popular Digium Digital Cards.

Need a T1 card with Echo Cancellation? Looking for a PCI Express card that has 4 FXO ports? Or, maybe you just need a Half-Length FXS card?

Whatever you’re looking for to complete your Asterisk® solution, Digium has the card that’s right for you. But act now – this offer is only good until August 29, 2014.

Why Digium Digital Cards?

Digium Digital Cards are the perfect, certified compliment to Asterisk® software in T1 and E1 environments.

But they’re also designed to be compatible with your existing software plus their open source drivers will support your custom API.

For more information, read more here:

Online Digital Card Selector

Click here to try the Digium Telephony Card Selector. Just scroll through the attributes to get matched up with the right card for your application.

Digium

Online Digium Telephony Card Selector

*Note:  Free shipping only applies to orders shipping within the Contiguous United States and where FedEx 2nd Day shipping is available. Free Shipping promotion cannot be combined with other promotions. This offer is good only until August 29, 2014.

VoIP Supply Becomes Xorcom Certified Dealer for CompletePBX Solutions

August 22, 2014 by Nathan Miloszewski
xorcom_voip_supply_cert_banner

Michael Taylor (center), VoIP Engineer at VoIP Supply Receives Xorcom Certification

We’re proud to announce that VoIP Supply is now a Xorcom Certified Dealer for Complete PBX Solutions.

Thanks to Michael Taylor (pictured above) our VoIP Engineer putting in all the hard work at the Xorcom technical training class.

We’re not sure what was harder for him the three days of in-depth training to learn all the details of Xorcom PBX installation, programming, and troubleshooting or being able to avoid all the distractions of the class location, Las Vegas.

If you’re not familiar with Xorcom the company was founded in 2004 and they focus on business telephony solutions for both VoIP and traditional PSTN. Xorcom products are based on Asterisk®, the open-source communication software used worldwide, for a flexible range of PBX solutions.

Taylor’s hands-on Xorcom training provided him with real-world application knowledge which means we that we not only provide you with the best first-level technical support but we can also help you:

  • Determine your communication needs and suggest the best Xorcom solution based on your line usage, infrastructure, and employee habits.
  • Streamline your phone system implementation process.
  • Ensure optimal phone up-time by securing the phone system.
  • Improve your overall user experience by providing communication efficiency suggestions.

Xorcom CompletePBX systems are pre-configured so they’re ready to use right out-of-the-box and they’ll give your single office/home office (SOHO), small and medium-sized business (SMB), or enterprise level applications lots of flexibility based on call management, Unified Communications (UC), and strong standard features.

Xorcom Case Study

How does a Xorcom PBX work in the real world?

This Chabot Space and Science Center case study is a good example of an application where cost and licensing fees were an issue. The customer was also going to install the solution themselves so they needed an Asterisk® solution that would work right out of the box and make calls straight away:

Extensive research into IP-based systems led Mr. dosRemedios to Asterisk®-based systems and he tried a test system, using trixbox CE and two Grandstream handsets. He liked what he was able to do, and so he looked for a turnkey system using Asterisk® at its core, and came across Xorcom and the Astribank concept. Because Mr. dosRemedios is the sole support for telephony at Chabot, his greatest concern was getting help configuring the system. So he contacted a local Xorcom reseller and soon realized that he could probably do everything himself.

Mr. dosRemedios chose a Xorcom XR2074 PBX with 1 PRI port, 8 FXO ports, and 16 FXS ports; with RAID (dual hard drive for redundancy), Rapid Recovery (for backup and restore
of the entire PBX), Yealink T20P handsets, and some premises cabling to fulfill the requirements. Deployment was performed in-house over a period of five weeks…The PBX installed without a hitch; Mr. dosRemedios connected four landlines to the FXO ports, and a couple of phones to the POE switch and was able to make calls the same day! He’s currently preparing to deploy the remaining 108 phones to run in parallel to the old PBX until the day he can move the PRI over to the Xorcom box.

Download the full case study here.

VoIP Q&A: EdgeMarc 250W Default Password and snom 300 Mounting Instructions

August 19, 2014 by Nathan Miloszewski

Questions about VoIP devices and services are regularly submitted to VoIPSupply.com through a technical support ticket or via the “Ask The Expert” tab on our product pages.

We respond to these requests directly but more often than not, this Q & A would be helpful for lots of other folks.

Below are your VoIP questions answered – Real questions from real people just like you.

Q: What is the default log in and password out of the box of the EdgeMarc 250W [Model] #250W-100-0002?

EdgeMarc 250W

EdgeMarc 250W Enterprise Session Border Controller Handles, and Protects, up to 10 Concurrent Calls

A:   The EdgeMarc 250W is an Enterprise Session Border Controller (SBC) that handles up to 10 concurrent calls, protecting them from malicious attacks. This SBC is designed for small and medium sized offices.

The default login is simply:

  • User Name = root
  • Password = default

Here’s a helpful screenshot from the EdgeMarc 250W Enterprise Session Border Controller Installation Guide:

EdgeMarc 250W

EdgeMarc 250W Default User Name and Password

Once you’ve programmed your EdgeMarc device you’ll want to change your login credentials.

Our friends at Bandwidth alerted us of a potential problem with passwords being compromised. Read more here:

In that post there are instructions on how to tell if your password needs to be changed.

Q: I am wondering if this phone [snom 300 UC Edition] comes with the parts to mount it to the wall or if I need to get a separate wall mount kit?

snom 300 UC Edition

snom 300 UC VoIP Phone Qualified for Microsoft Lync

A:   The snom 300 UC Edition is a basic 2-line VoIP phone qualified for use with Microsoft Lync.

To answer your question, the snom 300 comes with a dual-purpose footstand that allows you to either prop the phone up on your desk, or mount it on the wall with the pre-drilled holes.

Here are two documents and a screenshot (below) that you should check out:

 

snom 300

snom 300 Wall Mount

 

Stay Tuned

Check back next time for more VoIP Q & A.

Thanks for your questions!

Infographic: History of Analog Phones to VoIP

August 14, 2014 by Nathan Miloszewski

Our friends at Software Advice put together a great infographic highlighting the Life and Death of the Analog Telphone.

This pictorial history takes us through time from the humble telegraph to super-speed Voice over IP and beyond.

Designing the Analog Timeline

Craig Borowski, VoIP and telecommunications researcher at Software Advice shares some background:

The most interesting thing we learned while doing our research is the fact that the telegraph evolved into the telephone. It was fascinating to discover how that evolution took place. For example, as soon as the telegraph was invented, there were people all around the world who immediately started trying to improve it. The inventors kept making improvements in a similar fashion until it made the logical evolution into the telephone, and eventually modern-day VoIP technology.

The trickiest part with designing the timeline was how to conclude it. We felt fairly confident that the current trajectory of analog telephony points to a certain end — analog has been technologically obsolete for many decades. While we could have ended it at the present day, the FCC is showing signs of releasing big national carriers of their obligations to maintain analog systems. We felt the conclusion we chose was fitting — it really is only a matter of time. (Craig Borowski, Software Advice)

VoIP Spreading But People are Hard to Predict

Ben Sayers, CEO and founder of VoIP Supply, shares his experiences with the ongoing switch from analog to VoIP:

Ben Sayers

Ben Sayers, VoIP Supply Founder

Having been involved in telecom since 1994, I’ve seen a lot of the changes over the years.

My experience with VoIP began in 2004 when we launched VoIPSupply.com and began building VoIP solutions and helping others do the same.

Though I can’t predict the future of telecom and certainly wouldn’t put a date on it, VoIP continues to spread throughout the home and business world as it has reached a mainstream adoption phase.

There is little need for copper phone lines anymore, especially factoring in Mobile and all that it has changed over the last two decades.

With Skype and other desktop voice and video solutions having been around for such a long time without yet replacing the phone on most people’s desks, I have to expect that there is still a lot of runway remaining for fixed telecom with mobile integrations. People’s minds, habits and expectations are incredibly hard to change, including how they talk to other people. (Ben Sayers, VoIP Supply)

Here now is the infographic provided by VoIP and telecom review firm Software Advice

Aastra Discontinues 3 of Their 6700i Series Phones

August 13, 2014 by Tom Costelloe

So far 2014 has been a busy year for Aastra. In late January their acquisition by Mitel was completed and then in March they released the Aastra 6800i series of VoIP phones, which marked the first new line of desktop VoIP phones from them in quite some time. Now after a few quiet months some more changes were announced last week with the discontinuing of 3 phone models due to issues sourcing components.

Aastra 6700i

The Aastra 6757i, 6755i and 6753i were the cornerstone of Aastra and some of their most popular VoIP phones for close to a decade and with the utmost respect for these phones, I believe their time had come.

While them being put out to pasture may have been accelerated by the components issues (chipset) between the acquisition by Mitel and the release of the 6800i series phones I personally think it was a matter of when not if it was going to happen.

Again, I don’t mean this as a knock on the phones. I am fully confident that they will continue to serve their current users extremely well and would have continued to sell extremely well for Aastra; but, strictly from a line card perspective Aastra already had newer models out that not only supported newer, better, or more features but are also cheaper.

To help with the transition to alternative Aastra VoIP phone model we’ve pulled together some quick reference sheets that compare the discontinued models to those currently available from Aastra.

VoIP Supply Aastra 6753i Comparison Sheet

VoIP Supply Aastra 6755i Comparison Sheet

VoIP Supply Aastra 6757i Comparison Sheet

VoIP Q&A: Polycom SoundStation IP 5000 Conference Phone

August 7, 2014 by Nathan Miloszewski

polycom_ip5000Questions about VoIP devices and services are regularly submitted to VoIPSupply.com through a technical support ticket or via the “Ask The Expert” tab on our product pages.

We respond to these requests directly but more often than not, this Q & A would be helpful for lots of other folks.

Below are your VoIP questions answered – Real questions from real people just like you.

Q: How do you connect two calls to make a conference call [Polycom IP 5000] whether they call in or we call out? 

A:   The Polycom SoundStation IP 5000 Conference Phone is a 1-line SIP phone that allows you to create conferences with up to two other groups or, callers.

There’s a couple of ways to host a conference:

  1. Use the “Confrnc” soft key (top row of buttons with functions that display on the screen above)
  2. Or, when you already have an active call and a call on hold, use the “Join” soft key

After you set up the conference, you then have 3 options:

  1. Place the conference call on hold
  2. Split the conference call into two calls on hold
  3. Hang up and end the conference call and your connection

Now that the basics are out of the way lets get to your two scenarios.

Scenario #1: Both Conference Parties Call-In

You’ll be using the “Join” soft key in this example:

  • When the first caller calls you, place them on hold.
  • When the second caller calls you, press the “Join” soft key
  • You’ve now just created a conference call with the second caller (active call), the first caller (the held call), and yourself (you’ve been there all along).

Scenario #2: You Call Both Parties to Set Up Conference Call

You’ll be using the “Conference” soft key in this example:

  • Call the first party.
  • Press the “Confrnc” soft key. This places this active call on hold
  • Dial the number of your second party
  • Press the “Send” soft key.
  • When the second party answers, press the Confrnc soft key to join all parties in the conference

Need More IP 5000 Feature Info?

For all the detailed features and functions, check out the Polycom IP 5000 User Guide.

Don’t Talk Over My Shoulder – Echo Cancellation

What’s so great about the IP 5000, it’s been around for a few years now, you ask? One reason is the echo cancellation because, as this funny Polycom vid points out, you don’t want your calls to sound like someone’s talking over your shoulder:

IP 5000 Overview

For a more detailed overview of the IP 5000, watch our video here:

Stay Tuned

Check back next time for more VoIP Q & A.

Thanks for your questions!

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