Featured Article

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 services that include:

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 …

More from VoIP Insider

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:…

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?

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 …

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 CompletePBX Version 4.5

 

Xorcom Orion Video Conferencing for Small Businesses

Via Xorcom

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:

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 …

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?

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

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

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 …

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: 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:…

I recently wrote this post, VoIP Q & A: Phoenix Audio Speakerphone and Polycom RealPresence Group 300 Video Conferencing, in which I included some misinformation about the capabilities of the Polycom RealPresence Group 300 video conferencing system.

This question had been submitted to VoIP Supply:

“Is it possible to use the Polycom Realpresence Group 300 with online services such as Skype or Go To Meeting?”

I mistakenly wrote that “yes” you can use Skype and GoToMeeting with the RealPresence Group 300 because it is standards based.

Rookie mistake.

Thankfully Michael Graves of Graves on SOHO Technology called me out on it in his post, A VoIP Supply Q&A Batting .500!.

I was confusing the Polycom RealPresence CloudAXIS Suite with “standards based.”

Click here