7 Answers You Need Before Buying a Phone System

September 26, 2008 by Garrett Smith

Getting the right phone system for your business shouldn’t be difficult. After all, most of you who are reading this probably have better things to do with your time (like running your business or department). If it seems to you that phone system vendors do nothing more than confuse you with features, functions and technical jargon, you are not alone. Many folks throw their hands in the air when it comes to selecting a phone system and I can see why. Fortunately for you, it is not that hard to select the right phone system. No matter if you are a small medium business or a large enterprise, there is a basic set of questions you need to answer before selecting a phone system.

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Digium and Skype Partner

September 25, 2008 by Garrett Smith

Everyone was waiting for the big announcement out of Digium today and here it is…

Digium and Skype are partnering to bring Skype to Asterisk.

According to the official news, Skype and Digium are bringing a beta “Skype for Asterisk” add-on channel driver to market which will allow the integration of Skype functionality into Digium’s Asterisk software and enable customers to make, receive and transfer Skype calls from within their Asterisk phone systems.

More specifically, Skype for Asterisk will allow business users to:

  • Make, receive and transfer Skype calls from within Asterisk phone systems, using existing hardware.
  • Complement existing services with low Skype global rates (as low as 2.1US¢ per minute to more than 35 countries worldwide).
  • Save money on inbound calling solutions such as free click-to-call from a website, as well as receive inbound calling from the PSTN through Skype’s online numbers.
  • Manage Skype calls using Asterisk applications such as call routing, conferencing, phone menus and voicemail.

Free phone system + Free phone calls = a big win for businesses across the world.

Stay tuned for more information on how to participate in the BETA program.

Jazinga SOHO Phone System Hits The VoIP Supply Shelves

On Wednesday, our neighbors to the north, Jazinga, formerly released their new VoIP Phone System for the SOHO market. Now, I know you are probably thinking,

“Garrett, there are dozens of VoIP Phone Systems out there. What’s so exciting about another one coming to market?”

Well, I have had the privilege of using the Jazinga box in my home for the last few weeks and I can say that the Jazinga MGA120 is no “me-too” phone system.

Why?

Well for starters Jazinga is far and away the easy phone system I have ever set-up. The set-by-step wizard based set-up is a breeze – I even had my non-technical brother successfully configure the system in under 10 minutes – that’s impressive.

You can to this that it is SIP based, auto-configures most popular handsets, has WiFi, a four port switch, one FXO port for PSTN connectivity, full IVR capability as well as all of the essential call functionality you deserve from a phone system. This little box REALLY is the future of SOHO Phone Systems.

Fonality Goes Crazy, Releases HUD 3.0 For trixbox CE

Okay, so the folks at Fonality aren’t REALLY crazy; but if you have seen or played with HUD3, the third version of their Unified Microcommunications messenger you might think that giving it away for free is, well, a little crazy.

And that is exactly what they are going to be doing with HUD 3.0 for trixbox CE.

According to the folks at Fonality, the new HUD 3.0 will provide trixbox CE users with presence management and detection in a single interface for all types of office communications, including SMS, instant message, landline calling, mobile calling, chat, voicemail, email, conferencing, recording, and barging.

Now that is pretty exciting for businesses and call centers who are interested in making the switch to VoIP.

The driving force behind the decision to include HUD 3.0 in trixbox CE?

Bringing polished unified microcommunications to the open source community.

“Open source rarely lacks in features, but often lacks in ease-of-use and polish. Our intention with this announcement is to bring the polish of the HUD 3.0 unified communications platform, which is in use by more than 100,000 paid users, to the trixbox community. This should allow them, now more than ever, to compete with the high-prices of the big-iron oligopoly,” said Chris Lyman, CEO of Fonality.

HUD3 for trixbox CE should be available over the next few weeks.

More from: trixbox Garrett Smith

Introducing the Headset Compatibility Matrix

September 24, 2008 by Garrett Smith

Note: See the updated blog post: VoIP Phone and Headset Compatibility Guide is Now Available!

What Do You Know About Headsets?

Have you ever had the misfortune of purchasing a headset only to find out that it doesn’t work with your telephone, IP Phone, cellular phone or computer?

We have.

Ever find yourself wondering what headset best fits your business needs?

We have.

IP PHone HEadsets

You’ve probably asked yourself numerous questions like:

  • Do I want a corded or wireless headset?
  • Do I like the over the head or over the ear headset style?
  • Do I want monaural or binaural?
  • What is Noise Cancellation and how will it benefit me and my employees?
  • Do I need a handset lifter with my wireless headset?
  • Do I want Bluetooth or DECT (Digital Enhanced Cordless Technology) wireless technology on my wireless headset?
  • What headsets are compatible with my IP Phones?
  • If I go with a corded headset, what is the proper Quick Disconnect (QD) cord to work with my headset and IP Phone?

That’s why we have created the VoIP Supply Headset Compatibility Matrix for IP Phones – the answer to your Headset compatibility problems.

Headset Matrix For IP Phones

sheet1

(Click to Download Headset Matrix)

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Counterpath Announces X-Pro for Asterisk

According to Digium’s Steven Sokol, one of the primary organizers of Astricon, softphone gurus Counterpath…..makers of the popular X-Lite SIP softphone….have a new product release coming geared specifically toward Asterisk.

X-Pro for Asterisk

It’s called X-Pro for Asterisk. Details are sketchy at this point. I could find no mention of it on Counterpath’s website, other than the recently created X-Pro for Asterisk section on Counterpath’s forum.

More details as they become available.

Ask Mr. Andrews: Please Help, I’m a Newb!

September 9, 2008 by Garrett Smith

Q: Dear Mr. Andrews – I am new to Asterisk and VoIP enabled PBXes in general…can you point to any resources that will help me expand my knowledge?

A: The wonderful thing about technologies like Asterisk, FreeSwitch, trixbox and other “open source” platforms is the sense of community and opportunities for open exchange of ideas and general knowledge transfer that they foster.

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Truphone Wins Best For iPhone, Where Are The Others?

September 8, 2008 by Garrett Smith

The London Times has named Truphone, the “BEST (MOBILE VOIP) FOR THE IPHONE” Beating out other Mobile VoiP services such as Skype, Coms.net, NimBuzz and fring. Our own Cory Andrews reviewed the Truphone for iPhone application when it first hit the streets and while he had a bit of difficulty getting it to work at first, he reported that the application worked as advertised.

What I find most interesting about this article is not the fact that Truphone is a great service for the iPhone (it is), but the fact that there are not any other Mobile VoIP service applications out there challenging Truphone in the iPhone VoIP space is surprising. Sure there are a lot of Mobile VoIP applications for a jailbroke iPhone or Mobile VoIP applications that you can access via the iPhone’s web browser, but upon checking today, the Truphone application is the only iPhone VoIP application available through the iPhone Application store.

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Sangoma U100 Review

As Cory Andrews unveiled a few weeks ago, Sangoma has release a new USB FXO device, the U100, which allows you to turn a USB interface on an open source appliance/server into a two port PSTN connectivity device.
Sangoma U100 Review

Today we are going to tell you about our experience with the U100 (remember still in BETA, not all of the kinks have been worked out). To begin, this product shows some real promise in the residential and SOHO market. It’s hard to find a 2 line FXO product out there (Can anyone else think of an application for this nifty little device?). Pair this up with the new MSI Wind or the Shuttle X27 guzzling down no more than 40 watts of raw power and you’ve got an ultra green PBX on the cheap. Even greener would be getting it to run on the OpenWRT router that has USB ports (like ASUS WGL500). It’s actually kind of amazing something like this hasn’t filled this niche in VoIP to date.

Now, on to the good stuff…

The hardest part of the installation was getting the drivers to compile. I started with a stock Ubuntu Hardy Heron 8.04 with the 2.6.24-16-server kernel. Failed. Updated to 2.6.24-19. Failed. Finally, installed the 2.6.24.3 full source kernel. Failed. Through the entire process I was in contact with Sangoma developer Nenad Corbic who was extremely helpful in getting this thing to run in a very timely manner. Total props to Nenad and the team at Sangoma. A most impressive showing on how it’s done. We had gone from wanpipe drivers 3.0.5 to 3.0.7 to make this work, which is pretty good considering it’s really a beta version driver.

Once the drivers had compiled the rest was a breeze, as anyone who’s done a Sangoma install knows, they even setup the configuration files for you (ok not the dialplan, but please). Just reload Asterisk or FreeSwitch and you’re good to go.

According to the developers the U100 USB FXO does not have HWEC (hardware echo cancellation). But they do recommend using the excellent OSLEC software based echo cancellation or you can use the built in ones Asterisk uses. Just enable “echocancel=yes” in /etc/asterisk/zapata.conf. The configuration is the same as any other zaptel device in asterisk, so I won’t go into configuration files as that has been well covered in many howto’s.

Overall, this device was not bad to work with, even though it was still in BETA. As I stated above, this device nicely fills a niche in the SOHO/SMB space for PSTN connectivity. With some additional polishing up, you will undoubtedly see this device connected to many of the SOHO appliances in the future.

With that being said, can anyone have any thoughts as to how they would use this device?

More from: Asterisk Garrett Smith

How to obtain MagicJack SIP Credentials

September 5, 2008 by Garrett Smith

A VoIPInsider reader recently provided a tutorial on obtaining MagicJack SIP credentials, which should allow you to set up MagicJack as a trunk in any Asterisk based IP PBX by making the following modifications to SIP.conf. NOTE: VoIPInsider does not suggest, nor endorse activities which may violate your MagicJack TOS.

As of 5-31-08 to obtain your sip credentials you will need to dump your memory while magicjack.exe is running in order to view the decrypted password.
All other information can be had with any packet capture program.

Replace EXXXXXXXXXX01 with your MJ number. Include E and 01.
Replace the proxy proxy1.Atlanta.talk4free.com:5070 with the proxy your MJ registers to and change host=67.90.138.70 to host=YourProxyIPHere.
Replace XXXXXpasswordXXXXX with your password. Currently a 20 character string consisting of numbers and letters. Mine is all uppercase.

~~~~~sip.conf~~~~~

register => EXXXXXXXXXX01:[email protected]:5070

[magicjack]
context=incoming
username=EXXXXXXXXXX01
type=friend
secret=XXXXXpasswordXXXXX
port=5070
nat=yes
insecure=very
host=67.90.138.70
fromuser=EXXXXXXXXXX01
dtmfmode=inband
qualify=2000

~~~~~sip.conf~~~~~

~~~~~extensions.conf~~~~~

[incoming]
exten => YourMJNumber,1,Answer
exten => YourMJNumber,2,Dial(sip/sipura,30,r) ;dial someone…such as an ATA

[MagicJackOutgoing]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@magicjack,30,r)
exten => _1NXXNXXXXXX,2,congestion()
exten => _1NXXNXXXXXX,102,busy()
exten => i,1,Hangup
exten => t,1,Hangup
exten => h,1,Hangup

[sip]
include => MagicJackOutgoing

~~~~~extensions.conf~~~~~

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