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, 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.

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 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 offers a free basic account registration for their service.

Once you have created a basic account with 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.

ClueCon Telephony Developer Conference

April 24, 2009 by Garrett Smith

Are you an open source Telephony developer?

Maybe you’re aspiring to be one. Or perhaps you’re just a marketing or business professional that is intrigued by the possibilities presented by open source Telephony.

Regardless of who you are or your level of involvement with open source Telephony, you can take your knowledge to the next level by attending this years ClueCon Telephony Developers Conference!

What is ClueCon?ClueCon Telephony Developer Conference

Never heard of ClueCon? Well you’ll be kicking yourself for not learning about ClueCon sooner once you find out that ClueCon is an annual 3-Day Telephony User and Developer Conference bringing together the entire spectrum of Telephony from TDM circuits to VoIP and everything in between.

ClueCon presentations and sessions cover the following open source telephony platforms:

  • FreeSWITCH
  • Asterisk
  • Callweaver
  • OpenSIPS/Kamailio
  • Bayonne and Yate


Win one of three new IAX phones!

April 22, 2009 by Garrett Smith

A few days ago Insider Cory Andrews leaked information about a new VoIP phone that landed at VoIP Supply which supports the IAX2 protocol. For quite some time Asterisk lovers and open source nuts have been clamoring for a quality IAX compatible VoIP phone.

Well, today we’re proud to unveil the new Citel C4110 and announce a contest that could land you one of these bad boys for FREE (before anyone else) and a custom IAX GIAX (Eeks Geeks) T-Shirt!

Before we get to the contest, let’s look at the C4110.

The Citel C4110 is a stylish business class VoIP phone with two line appearances, dual Ethernet ports, Power over Ethernet, SIP and IAX2 protocol support. The C4110 can also be configured via a web gui or auto-provisioned using TFTP.

All of this for only $99 USD! (Plug: We are now accepting pre-orders. Supply is limited.)

And let’s not forget the IAX GIAX (Eeks Geeks) T-Shirt:

With that out of the way, it’s on to the contest.

The theme of this IAX phone contest is, “Why do you use IAX protocol?”

In order to enter the contest, you must do one of two things (or both if you want two chances):

  1. Leave a comment below about why and or how you use IAX protocol in your VoIP deployment(s).
  2. Write an article on your website, post on your blog, tweet on your Twitter or post a note on your Facebook about the differences between IAX and SIP with a link back to this contest.

This contest will run until Monday, April 27th at 5pm EST. At that time, the Insiders will meet and pick three winners (probably from a hat).

So if you’re interested in getting your hands on an FREE Citel 4110 before others can even buy it and your custom IAX GIAX T-Shirt spend 5 minutes of your time leaving a comment below about why and how you use IAX protocol or use your own site to let the world know about the differences between IAX and SIP!

More from: Asterisk Garrett Smith

5 Ways to Minimize the "Hassle" of Switching to VoIP

April 20, 2009 by Garrett Smith

Earlier today Doug Mohney wrote a piece at FierceVoIP about the resistance to migration many cable providers are facing from business customers. In the word’s of one cable company exec, “Nobody wakes up and says, ‘Today I’m going to change all the phones in my business.”

While this may be true, switching to VoIP isn’t a hassle.

At least it doesn’t have to be, since here are a number of things that businesses (and those serving them) can do to minimize the hassle of switching to VoIP.

Follow the VoIP Insiders on Twitter

April 17, 2009 by Garrett Smith

Unless you’ve been living under a rock for the last year you’ve heard of Twitter.

If you’re one of the unfortunate few who happen to live under a rock Twitter is a great way to waste time micro-blogging platform that allows people to have conversations. You really should check it out if you haven’t already.

Anyways, if you’re on Twitter you should consider following us. Right now we’ve got three very active Insiders accounts:

@voipsupply – This Twitter account features VoIP news, resources and advice from the team at VoIP Supply. It also showcases the occasional deal or two…

@coryandrews – This is the Twitter account for resident VoIP product nut Cory Andrews. Cory Tweets about a large variety of subjects in the technology space.

@garrettsmith – This is my Twitter account. You’ll find more VoIP related information and advice, along with occassional quips on business and marketing.

We love connecting with readers, customers, vendors and those within the VoIP industry, so if you’re on Twitter consider following us!

Labs: Setting Up the Polycom VVX 1500 with Switchvox

Earlier this week we unboxed the new Polycom VVX 1500 Video Phone and promised to share some details of our experience in integrating the VVX 1500 with Switchvox SMB 4.0, the IP PBX that we currently run in production at

Getting the voice side of things up and running was easy, but we quickly ran into a few issues while trying to make video calls. The Polycom VVX 1500 is currently not supported on the Switchvox platform by Digium/Switchvox or Polycom. There are known issues with the VVX 1500 on Switchvox and both Digium/Switchvox and Polycom are working toward being able to fully support the product.

Being impatient types, we rolled up our sleeves to see if we could come up with a workaround. Shame to have the VVX sitting on my desk and not being able to utilize video calling. Big thanks to our head product engineer and VoIPSupply Labs mad-scientist-in-residence Chris Heinrich for coming up with the following solution. (more…)