VoIPSupply Labs: Integrating Nokia "E" Series Phones with Switchvox PBX (Plus Asterisk or any SIP IP PBX)

May 12, 2009 by Garrett Smith

Do you already own a Nokia “E” Series mobile phone, or are you considering purchasing one? The Nokia “E” series phones, although not inexpensive, offer the flexibility of both GSM and WiFi/SIP calling, and can be integrated with most SIP based IP PBX platforms. For the purpose of this article, we chose a Nokia E51 and configured it to leverage any open WiFi connection to register with our Switchvox IP PBX.

These instructions apply to and of the Nokia “E” series dual mode GSM/WiFi mobile phones.

NOTE: The SIP registration instructions below assume you have already setup your WLAN settings on the phone, and will use WiFI to communicate to your SIP server. It is always suggested to have the AP on the same network as the IP PBX and to place a certain level of security.

1. First, power on the Nokia “E” series phone by holding down the power button located on the top of the phone. (more…)

POTS lines are an endangered species

May 11, 2009 by Garrett Smith

I know the title of this post is telling you something that you probably already know. Unfortunately, not everyone is as informed as you are.

Case in point, Larry Midge at Cnet.

Larry is worried about land lines. His kid’s prefer their cellphones. He even bought one of them Vonage – which they never even used.

And apparently Larry’s kids aren’t alone. Recent research suggests that 20% of all homes in the US are now “cellular only”.

You don’t need Larry’s story or research to know that POTS lines are a thing of the past. Unless you’ve grown-up with one or don’t know alternatives exist, your not likely to look into a POTS line if you need voice service.

But just because POTS lines are declining in popularity and at one point will be extinct, they still have their place.

As Larry pointed out, neither cellphones or VoIP service touts the reliability of a land line. While cellular and VoIP are inching closer in reliability to POTS lines, if the power or network goes out, you won’t be able to use them. Unlike your POTS line.

That’s why it’s recommended by the folks here at the VoIP Insider that residents and businesses switching to VoIP consider keeping a POTS line (or more) for fail over purposes.

In most cases the price points for VoIP service is such that one keep a basic POTS line ($9.99 – $15.99 plan) that provides per minute dial tone (or local calling)still provides considerable monthly cost savings. This is especially true for businesses.

So yes Larry, POTS lines are an endangered species. But at least they’re not extinct.

First Look: Blue Microphones Eyeball

May 6, 2009 by Garrett Smith

Last week I got the opportunity to get out of the office and head down to beautiful Boca Raton Florida for a retailing conference. While I was there I got the opportunity to check out a ton of new products and services that are either new or coming to market.

One product that caught my eye (no pun intended) was the Eyeball from Blue Microphones. Unless you’re a studio musician or into voice recording you’ve probably never heard of Blue Microphones. After all, they’re best known for making high-end microphones for likes of Cold Play.

I’m no lead singer, but I do use many VoIP soft clients throughout the day. So when I found out that they had an HD voice and video webcam, I was excited to get my hands on it.

Lucky for me, the great guys over at Blue Microphones stuck a demo unit in my bag. Unfortunately I’ve yet to play with the Eyeball (more on that when the review hits later this week), but from what I saw down in Boca, the Eyeball touts:

A very cool retro design
HD voice
HD video with performance up to 30fps
Ultra portability through an integrated stand and case
Easy mounting

The Blue Microphones Eyeball has a MSRP of $139.99, so it’s definitely more expensive than your run-of-the-mill webcam. Considering the promise of HD voice and video there might be a case.

But will HD voice and HD video all wrapped into a slick retro wrapper be enough to turn me into a voice and audio snob? Check back later this week for the answer (and a full product review)!

First Look: New Cyberdata VoIP Intercoms

May 5, 2009 by Garrett Smith

Amongst the most innovative vendors in the VoIP product space, Cyberdata understands business communications problems and designs their products with customer needs foremost in their minds.

Cyberdata has just released two new SIP intercom products compatible with most SIP-based IP PBX servers that comply with the SIP RFC.

First up is their new indoor intercom. The CyberData SIP-enabled VoIP Indoor Intercom is a Power-over-Ethernet (PoE 802.3af) and Voice-over-IP (VoIP) door entry device that easily connects into existing local area networks (LANs) with a single cable connection.

Next up is their Indoor Emergency intercom. The CyberData SIP-enabled VoIP Indoor Emergency Intercom is a two-way communications device that is used in an area where either an emergency panic button or two-way priority communications are required. The intercom is compatible with most SIP-based IP PBX servers that comply with SIP RFC 3261.

Look for both products available soon for purchase on VoIPSupply.com

Gartner tells us what we already knew about Mobile VoIP

by Garrett Smith

This morning the Gartner group dropped some hot new knowledge on us about the growing threat mobile VoIP presents to incumbent carriers.

Wait. That’s nothing new. We’ve known about that for years.

Gartner’s announcement did shed light on what they consider the actual market opportunity for mobile VoIP services to be, however, as well as where these where all these mobile VoIP minutes will be originating from.

According to Gartner research director Akshay Sharma, in ten years more than half of mobile voice traffic will be carried end-to-end using VoIP. That’s approximately $345 billion dollars worth of minutes.
Of this mobile voice traffic, 30 percent will be driven through third parties such as Google, Facebook, Yahoo and others who will look to add voice as a value added feature to their current offerings.

These numbers are a little misleading if you’re a mobile VoIP provider salivating over the market opportunity these next eight years.

Incumbent carriers are already leveraging VoIP behind the scenes and will likely ramp-up their efforts to provide mobile VoIP services themselves once third party services from companies like Truphone and even Skype eat away at their voice revenues – cutting into the opportunity for third party players.

This is fine by many current mobile VoIP providers such as Mobivox and Nimbuzz who have positioned themselves as value added platforms for carriers – not direct competitors.

Regardless of how the players are shaping up, mobile VoIP still has some technical hurdles preventing it from realizing its full market potential. Many users are still struggling with the awkward nature of many of the services – something that will only improve with time and better technology.

Until that time mobile VoIP will continue to hold promise. But you already knew that.

SIP Trunking Redux

May 4, 2009 by Garrett Smith

Back in October 2008 we featured a piece on SIP Trunking, which explained the basics of the technology and the potential benefits for business users.

SIP Trunking continues to proliferate, and I recently came across some excellent stories related to SIP Trunking that I thought I would share.

Over at NoJitter.com, Alan Percy from Audiocodes explains the difference between BYOBB (Bring Your Own Broadband) and Bundled SIP Trunking offerings.

Also on NoJitter you can find a comprehensive piece on SIP Trunking from Matt Brunk, detailing some of his personal experiences with the technology.

Gary Kim at the IP Carrier Blog postulates on the prospective growth of the SIP Trunking market between now and 2103.

TMCNet’s David Byrd talks about the impact of bandwidth metering on SIP Trunking.

Cisco Interaction Network blogger Robb Boyd wonders, is SIP Trunking the next big thing?

There’s no question people are talking about SIP Trunking. They’re also Tweeting about it. If you have thoughts on SIP Trunking, pros, cons, personal experiences, predictions, etc….we’d love to hear from you.

And the IAX phone winners are…

May 1, 2009 by Garrett Smith

Last week we launched a contest here on the VoIP Insider that gave you the opportunity to win a brand new Citel C4110 IAX phone by answering the question, “Why do you use the IAX protocol?”

In total there were over 100 contest entrants. From comments on the post, blog posts, tweets and even Facebook notes, the word definitely got out about the reasons and benefits to using the IAX protocol.

After picking names from a hat, the winners of a brand new Citel C4110 IAX phone and an IAX GIAX T-Shirt are:

Steven Johns (Post comment)
James Finstrom, @geek3point0 (Tweet)
Ruben Olsen (blog post)

Winners will be contacted soon about claiming their prizes.

Now, if you’re not Steve, James or Ruben, but participated in the contest, you can still be a winner.

Tech Tip: Programming a Hold Button on the Snom 300

April 29, 2009 by Garrett Smith

Programming a Hold Button on the Snom 300

The Snom 300 has everything you could ask for in a budget-minded, business class SIP desk phone…..Dual RJ45 Ethernet Ports, 2-Line LCD Display, Power Over Ethernet and G.729a support. At a street price of $119.95, the Snom 300 represents a tremendous value.

The only gripe I have heard levied against the Snom 300 is the lack of a pre-programmed “hold” key on the phone. Turns out, this is an easy fix….thanks goes out to Tom Ostrander, Eastern Regional Channel Manager at Snom for sharing the workaround with us.

Here is how you would program one of the function buttons into a hold button on a Snom 300:

1. You can reprogram any of the following four buttons on the Snom 300 into a hold button: Redial, Directory, Transfer, Mute (you can also change Line 1 & Line 2 but I don’t know why you would want to do that).

2. Go to the web GUI for the phone (by typing the IP address for your Snom 300 into the web browser on your computer)

3. Go to Function Keys

4. You will make the change under the column for number, the change will be to F-R for whichever key you want to change to hold. For example if you want to change the Directory key to the hold button you would change F_ADR_Book to F_R

5. PRESS SAVE. The directory button is now a hold button and you can re-label it if you would like to.

Update: Citel C4110 IAX Contest

by Garrett Smith

Last week we posted a contest centered around the new Citel C4110 IAX Phone.

We were originally going to pick (3) lucky winners on Monday, April 27th at 5pm EST. Due to an unprecedented amount of response, we have extended the contest through this Friday, May 1st. We will be awarding (3) Citel C4110 phones and (3) IAXGIAX shirts to folks who commented about, or linked to, the story.

First Look: SIPDroid Open Source SIP Client for Android Mobile Phones

April 28, 2009 by Garrett Smith

SIPDroid is a java based, open source SIP client that has recently been developed for use with mobile devices based on Google’s Android platform.

Based upon a Java SIP stack contributed by MJSip, SIPDroid is currently in public beta.

The SIPDroid Users forum can be found here. The SIPDroid Developers forum is located here.

From the SIPDroid.org website:
After completion of the closed alpha stage this project will publish the software for free under the terms of GNU General Public License v3. The first beta version will be for software testing. So please allow for some issues and incompatibilities at the beginning.

Although SIPDroid will likely mature quickly, it is currently only fully supported using virtual PBX service from PBXes.com. PBXes.com offers a free basic account registration for their service.

Once you have created a basic account with PBXes.com you can set up additional SIP providers/registrars within the Trunks section of their web based UI.

PBXes allows you to register several trunks from multiple telephony service providers of your choice. PBXes routes incoming calls over SIP and the PSTN to you. If you are online you can take a call as VoIP, and if you are offline the call will fall back to GSM.

Beyond their free basic service, a paid account additionally allows for handoff of calls between networks. PBXes also supports NAT.

To install Sipdroid you need version 1.5 “Cupcake” of Android. It is already available from HTC for Android Developer Phones. Visit this link for details on updating the OS. An OTA (over the air) update for the other phones has been announced for coming in the beginning of May.

Tragically, most of us here at The VoIP Insider are Apple iPhone users, but we have procured an Android mobile phone from Yannick Tessier, our head of engineering, for testing purposes. We will attempt to get SIPDroid working this week with Asterisk and let you know how we fare.

VoIPSupply Labs: Integrating Nokia "E" Series Phones with Switchvox PBX (Plus Asterisk or any SIP IP PBX)

POTS lines are an endangered species

First Look: Blue Microphones Eyeball

First Look: New Cyberdata VoIP Intercoms

Gartner tells us what we already knew about Mobile VoIP

SIP Trunking Redux

And the IAX phone winners are…

Tech Tip: Programming a Hold Button on the Snom 300

Update: Citel C4110 IAX Contest

First Look: SIPDroid Open Source SIP Client for Android Mobile Phones

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