Grandstream GXV3140 Now Skype Certified

As Skype continues to evolve as a service provider it’s inevitable that they will continue to find ways to move away from the desktop and onto the desktop.

Over the summer, Grandstream released a new firmware version that allow it’s GXV3140 to natively support Skype and Skype video calling. Today, Grandstream has taken this one step further by announcing that they Grandstream GXV3140 is now Skype certified.

What’s this mean?

Well, unlike this summers announcement, it means that Skype has officially given the Grandstream GXV3140 its blessing, making it the second videophone on the market for use with Skype. And according to Grandstream,

“Skype certification on the GXV3140 IP multimedia phone creates a seamless video-enabled communications experience from PC to desk phone, giving Skype users the utmost flexibility and convenience for video calling. Grandstream’s GXV3140 bridges the gap between convenient-to-use dedicated voice/video end-points and the increasingly popular world of PC-based Internet multimedia communication offerings like Skype.”

The idea of seamless video communications is a major hurdle to widespread video calling adoption. For the most part there is little interoperability between video calling services and hardware, leaving most companies interested in video calling hesitant to pull the trigger for fear of living in a walled garden.

While this announcement certainly won’t change this, it is a step in the right direction and a sign that more seamless video communications solutions are coming.

Switchvox Mobile on your iPhone or Blackberry

Are you mobile?  Do your clients have to juggle 2 or more phone numbers to reach you because you are sometimes at your desk but most of the time all over the place?  Don’t you wish your business card were a little less cluttered with just one contact phone number that you can be reached at all the time?  You can have these things now, and you probably won’t even have to buy anything!

Businesses running Switchvox 4.6 or 4.5 already have the ability to stream the Switchvox desktop phone functionality to iPhones and Blackberries.  If you are using Switchvox for your desktop phone system and you have an iPhone or a Blackberry, you are less than 10 minutes from enjoying all the features your desk phone has to offer on your SmartPhone!

Ready? Let’s get mobile!

Once you enter the App store, search for Switchvox Mobile.  This App is a FREE DOWNLOAD!!!! (Don’t you just love FREE)

Switchvox Mobile for Blackberry

-          For Blackberry go to www.BlackBerry.com/AppWorld 

-          For iPhone go to www.apple.com/iphone

(This is neat…. If you don’t have Switchvox yet, but want to see how Switchvox mobile works, you can download the App and use it in DEMO mode!!!! )

-iPhone Users – click here for easy to follow instructions on how to set up your iPhone for Switchvox Mobile (with Pictures!)

-BlackBerry Users – click here for easy to follow instructions on how to set up your Blackberry for Switchvox Mobile

See details, listen to your voicemails, and return calls at the touch of a button.Access your Switchvox Company directory without logging into the internet.Remotely set your greetings without a chain of touch-tone commands.Make calls out either from your SmartPhone number or your Switchvox extension.Set up call rules using an easy GUI while away from the office.See call history from your SmartPhone while nowhere near your desk. Even see call history on your SmartPhone for calls you made from your desk!

SOHO DECT Systems

DECT Solutions for your SOHO Environment

So after reading my first two posts on DECT technology and how to design a DECT system, you are now ready to move forward with a solution that fully meets your needs from a hardware perspective. There are a few solutions that are viable to meet your needs with more on the brink. Since DECT phones are becoming more popular by the minute in the VoIP industry, we are seeing manufacturers entering this market and delivering robust cordless solutions that can meet the needs of any SOHO application. Below, we will detail a few of these solutions.

The first solution is actually a new solution and actually a new “player” in the SIP DECT world. Panasonic who is well known for their analog cordless handsets which are very popular in your large retail chain stores such as Best Buy, Target, or Wal Mart have introduced 2 cordless SIP DECT Solutions to the market. The first solution is the KX-TGP550 which is a corded desk IP phone that ships with 1 additional DECT handset which pairs to the base phone. If you are familiar with the Aastra CT solutions, this is very similar EXCEPT, the KX-TGP550 can supply up to 6 additional KX-TPA50 DECT handsets and EACH handset can be allocated as its own individual unique SIP extension unlike the Astra solution that only allows 4 handsets and each handset inherits Line1 configuration of the base phone. Here is a good look at how the KX-TGP550 and KX-TPA50 handsets function.

The solution allows for 8 individual SIP extensions but SIP extensions may also be shared between each handset and/or base unit. A good look at how this is configured via the web GUI interface is below:

The KX-TGP550 base phone can also act as BLF/Speed dial application as it contains 6 keys which are each dedicated to a unique handset paired to it. Receptionists can use these buttons to see if the person is on a call or available to take a call and even press that users handset button to speed dial them. They can also press a “ALL” button which essentially calls every handset on the DECT network.  A good look at how this is setup and displayed is below:

The next solution offered by Panasonic in the SIP DECT world is the KX-TPG500 which offers a DECT base station that physically connects to your LAN network via cat5 and transmits DECT signal as well handles the SIP registration between the paired DECT KX-TPA50 handsets and the SIP Call server or provider. The KX-TPG500 ships with a single base station and handset. It can support a total of 6 DECT KX-TPA50 handsets and 8 SIP registrations. Here is a good look at how this solution is setup:

Both solutions are viable for SOHO applications and can completely transform your home or small office into an entire cordless mobile phone system. The only downside to these solutions is that there are currently no DECT repeaters available to extend DECT signaling to greater ranges.

The next DECT SOHO solution is manufactured by Snom and has been around for quite some time. The Snom DECT solution is called the Snom M3 . The M3 ships with a DECT base station that transmits DECT signal and also handles SIP registration between the 8 additional M3 handsets that can be paired to it. It ships with 1 handset and each handset can be programmed with its own unique SIP extension. The base can handle 3 concurrent calls at any time and unlike most DECT solutions, the M3 DECT solutions offers DECT repeater options that can be spread out within the main base’s DECT frequency to offer a greater range of DECT signal coverage. The base can support up to 6 additional DECT repeaters which can be purchased separately. Below is a good example of the Snom M3 solution and how the DECT repeaters offer greater signal coverage.

Each Snom handset supports a speakerphone option, 2.5mm headset jack, customizable ring/vibrate options, color LCD, out of range notification, low battery notification, and ships with its own desktop charger.

Aastra also offers a SIP DECT Solution called the MBU-400. The MBU-400 ships with a single MBU-400 DECT base station that broadcasts DECT frequency and handles 8 unique SIP registrations between each paired Aastra DECT 420d handset. The MBU-400 ships with 1 420d handset and can handle up to 8 total 420d handsets which can be purchased separately. The base is unique to other DECT solutions because it can handle 4 concurrent calls, 3 VoIP over the LAN connection, and 1 PSTN via its FXO port. This is also useful in cases where your network may go down. The FXO port is also utilized as a lifeline and calls are automatically routed over to the FXO port if the base loses its SIP registration or network connectivity.

Each 420d handset offers a speakerphone, 2.5mm headset jack, ring and vibrate options, out of range indications, low battery indications, ships with its own desktop charger, color LCD, distinctive ringing, customizable phonebook, and over the air updates. The downside to the Aastra MBU-400 is it doesn’t offer any DECT repeater option.

And last but not least, we have Siemens Gigaset which offers 2 SOHO SIP DECT solutions, the A580IP and S675IP. Pictured below the A580 IP ships with (1) A58H DECT handset and 1 DECT base station that interfaces with your LAN network. The base can support up to 6 total Siemens Gigaset DECT handsets and Siemens provides a few different models linked and pictured below. All handsets support Wideband voice quality through HDSP technology which is a feature that sets this solution apart from any of the DECT solutions talked about previously.

Additional Handsets that are supported on the A580 IP solution and can be mixed and matched include the following:

The second SOHO DECT solution offered by Siemens Gigaset is the S675 IP. The S675 IP solution ships with 1 S67H DECT handset and like the previous solution can support up to 6 total DECT handsets paired to the S675 IP base unit. The base unit contains a single LAN port to connect to your network as well as 1 FXO port to connect to your local land line. It supports 3 concurrent calls, 2 VOIP and 1 through the FXO port via PSTN. This is also used for failover. Below is a picture of the S675 IP and a listing of its supported HDSP DECT handsets.

Additional Handsets that are supported on the A580 IP solution and can be mixed and matched include the following:

And that about does it for SOHO IP DECT solutions. Stay tuned for my nest post which will describe a few enterprise level DECT solutions.

VoIP Failover with Xorcom Twinstar

Introducing Twinstar…what do you know about Failover!!!

If you are new to the VoIP game or are a seasoned veteran, we all know that in business…we need to be able to make and receive phone calls. Most businesses rely on this type of communication to handle the day to day business activities. Also, it is required in most cases of an emergency.  With the introduction of IP PBX’s and VoIP, we all know that there are great ways to reduce costs as they relate to your recurring  traditional phone “bills”, but what happens if your  IP PBX decides it wants to crash and fail. In either case, this is a recipe for disaster. What if you can’t make phone calls? Can you run your business? How much down-time can you afford? These are all questions you must ask yourself and consider when deploying a VoIP solution…because these are all very real scenarios that have happened and could happen to you, that is, if you don’t protect yourself.

Recently, I had the pleasure of attending a Xorcom hands-on technical training in Las Vegas, NV. Yes, I actually did pay attention in class and wasn’t distracted by those chirping slot machines just a short walk down the hall. If you don’t know of Xorcom, or they are new to you, you can find more information about them here.

During the training, Xorcom demonstrated the 1st 100 percent reliable complete failover solution, they call Twinstar. Twinstar is Xorcom’s method of delivering a complete failover solution without any personal involvement. That means that you don’t have to get out of bed at 4am to cater to an alarm that your IP PBX is having issues. Rest assured, with the solution below, you will be 100 percent covered when it comes to making phone calls. Trust me, I seen it in action and you will too.

The Xorcom Twinstar solution comprises of (3) required hardware components and (1) software requirement. First, you need to have the following:

-A Xorcom Astribank. An Astribank is a USB peripheral device that connects to the PBX via USB. The Astribank will allow your VoIP solution to connect to the outside world, either via FXO PSTN, or Digital T1. The Aastribanks are modular and can support PRI signaling either via T1 or E1, and FXO/FXS PSTN RJ-11 connections. You will want to ensure you also suit your Astribank with the dual USB ports for Twinstar failover Support. You can check out the lineup of Astribanks here.

- (2) Xorcom XR2000 or XR3000 IP PBX’s. Xorcom manufacturer’s three models of IP PBX appliances, two of which are suitable for Twinstar failover. The Xorcom XR2000 appliance supports up to 200 users, 60 concurrent calls, with 1GB of RAM and optional RAID support on its hard drives. More details on the XR2000 can be found here. The XR3000 is Xorcom’s “big daddy” server that can support up to 1000 users, 300 concurrent calls, with 1GB of RAM with the option to upgrade to 4GB, optional RAID support on its hard drives, and an Intel Core 2 Duo processor with the option to upgrade this as well, to a Quad Core processor. More information on the XR3000 can be found here.

- Xorcom Twinstar LC0016 – This ensures your Astribanks come with the required 2ND USB Port and necessary USB cables to connect to your Xorcom XR PBX, to support Twinstar. This also ensures your Astribanks come with the necessary firmware to support Twinstar.

- Complete PBX or Xorcom’s Elastix based Software- Lastly, in order to take advantage of the Twinstar failover solution, both XR appliances must be pre-loaded with Xorcom’s Complete PBX or Xorcom’s Elastix based software. The Elastix-based software is free of charge. The Complete PBX version is paid commercial software developed and offered only through Xorcom directly. When you are talking to your VoIPSupply sales rep, tell them that you want either Xorcom’s Elastix based software or their Complete PBX software loaded. If you would like the Complete PBX software, we can have a Xorcom representative contact you immediately after your hardware purchase through VoIPSupply to discuss pricing and feature support.

Now that we have covered your hardware and software requirements, let’s talk about how the solution works.

Step 1:

Let’s start with the Astribank suited for twinstar failover with dual USB ports. You will first connect your PRI (T1/E1) connections, PSTN RJ-11 FXO connections, and any analog telephones to your RJ-11 FXS connections on your Astribank. This will satisfy your means of making and receiving calls to and from the outside world.

Step 2:

Setup both of your XR servers whether they are the XR2000 or XR3000 with the SAME IDENTICAL ASTERISK CONFIGURATION. No my caps lock is not stuck. This is very important as it relates to the failover portion. Remember, your XR servers will need complete PBX software. Once you have loaded the identical configuration to each server, you will want to designate 1 server as the Primary server and the other as the secondary. Both servers will be on and running simultaneously, but we need to tell the Astribank which one is live (primary) and which one is the backup (secondary). Also, keep in mind, when you make an asterisk configuration change on the main server, you will want to copy or replicate those changes to the secondary server. You should do this daily as you may never know when an incident or problem may happen. You can do this via DRBD Synchronization. More on this setup can be found in the technical guide linked below.

Step 3:

Connect your primary XR server to the Astribank’s USB (MAIN) Port. Then connect your secondary or backup server to the 2nd USB Port or the (BACKUP) Port as pictured below.

Now that we have gone through steps 1-3, your hardware solution should look something like this:

Step 4:

OK, now this is where the configuration may get a little technical. Within the asterisk configuration, you will create a cluster IP address that encompasses both of your XR servers. For instance, my primary XR server’s IP address will be 192.168.1.2, my backup XR Server will be 192.168.1.3 and the cluster IP address will be 192.168.1.1. The cluster IP address encompasses both XR servers which you will use to register your IP Phones to. This allows your IP phones to easily failover if a problem is detected. I am aware of certain IP Phones support a backup SIP server option in their configuration but not all manufacturers do. In the twinstar failover solution, you can use any IP phone you want, just ensure you register them to the cluster IP address. Here is a small look at this setup:

Step 5:

Test the Twinstar Failover solution- Once steps 1-4 are complete and you are properly making and receiving phone calls from your IP phones or analog phones connected to Astribank, pull the power plug on your primary server. This will cause twinstar to take action and immediately failover all connections including PRI (T1 or E1), FXO or FXS, SIP Trunks, and your SIP phones to the backup server. This failover switch takes about 10-20 seconds to complete but requires no unplugging and movement of cables or equipment. After the switch, you are now back up and running on a fully functional phone system identical to the one that just failed without a hiccup…well there is one hiccup, if you were on a call during that transition to failover, your call will be dropped. This is next step for Xorcom, to achieve, a fully redundant failover solution without any in-call interruptions.

FreeSwitch PBX

FreeSWITCH™ is an open source communications platform. FreeSWITCH™ is a library which ships with a small executable that loads the library, launches the core, and performs the various tasks that are defined by the modules. In its base form FreeSWITCH™ is a soft-switch or PBX telephony application, not completely unlike Asterisk but capable of handling thousands of simultaneous calls.

FreeSWITCH™ makes it possible to build a softphone, PBX system, soft-switch, or interface with other open source PBX systems such as OpenPBX.org, Bayonne, YATE or Asterisk. It can also be used to build a VoIP switching platform uniting various technologies such as SIP (using the Nokia Sofia library), H.323, IAX2, LDAP, Zeroconf, XMPP / Jingle etc.

As a library FreeSWITCH™ can be used by developers to enable switching in their custom applications. FreeSWITCH™ can be launched not only from a C application, but also via PHP, Perl, or a variety of other languages.

FreeSWITCH™ is written in C, built from the ground up (not a fork of another code base). It is designed to take advantage of as many existing software libraries as possible. It has a modular, extensible architecture, with few and necessary functionality in core ([libfreeswitch]]) with optional modules to do the rest.

FreeSWITCH runs on Windows, Mac OSX, Linux, *BSD, ARM, and other Unix flavours.

It is licensed under the MPL.

• Runs on Win32/Mac/Unix
• IVR API
• 8kHz/16kHz/32kHz/48kHz audio
• Soft conferencing
• SIP B2UA/SRTP/TLS
• SIP BLF/SLA/PBX features
• Presence
• GoogleTalk
• IPv4/IPv6
• ENUM/ISN
• Async audio
• Event/logger engine
• Real time
• zRTP (libzrtp) •SIP (Sofia-SIP)
•IAX2 (LibIax2)
•GoogleTalk (dingaling)
•H.323 (OPAL) (beta)
•Skype (SkypIax) Watch 4 part video of Giovanni Maruzzelli on SkypIax.

Asterisk Village Telco

an easy-to-use, scalable, standards-based, wireless mesh, local, do-it-yourself, telephone company toolkit

Website: http://www.villagetelco.org/

A project to bring low cost telephone service to rural areas using Asterisk and a WiFi mesh network.
The goal is to provide a complete solution for communities and Villages to take charge of their own telecommunication needs.

The MeshPotato includes mesh network (batman) and Asterisk in a weather proof casing and using only 2-3 watts. It can be runned on a small battery and a solar panel. The Project is sponsored by the Shuttleworth Foundation.

AstriCon 2010: Day Two-The Asterisk Party

The spotlight for the 2nd day of AstriCon 2010 is the Code Zone Party held in BOBBY McKEYS. It’s a great party for the Open Source communty members. We drink, we talk,we share and we laugh. The key difference this year of the party is that everyone knows that Mark Spencer is not only good at writing code but also playing piano!!

Mark Spencer is Playing the Piano!

Belief Mo & Mark Spencer