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Asterisk

Asterisk Linux Based IP-PBX

Gizmo5 SIP Trunks available in trixbox CE

September 26, 2008

Gizmo5 SIP trunks have always been available in trixbox CE, but it was a manual process. The Gizmo5 team has built a module to be part of the trixbox package manager that allows you to purchase your trunks, see your account balance, purchase more minutes, and automatically setup your inbound and outbound routes. The module is now available via the trixbox package manager and will be built into all upcoming ISO builds.

Additionally, the calling service for trixbox CE is pre-configured to use the Gizmo5 calling network and includes a new UI for easy administration.

More on Skype for Asterisk

September 25, 2008

Continuing the coverage of the big Skype for Asterisk news I covered earlier today... In a nutshell, the Asterisk server acts as a Skype-to-SIP gateway, a very popular requested feature, mapping Asterisk SIP-based phones onto the Skype network via the Asterisk Skype channel driver. Technically, you could call Asterisk a Skype-to-IAX gateway as well.

So how does it work?

Well, on an inbound call to your Skype username, both your Skype desktop client rings (if running) and your Asterisk IP phone rings. You can take the call using either your PC's Skype software or your IP phone.



Skype for Asterisk Launches

September 25, 2008

Skype and Digium have hooked up to bring Skype to Asterisk called Skype For Asterisk. Skype For Asterisk launched minutes ago enables Asterisk users to get access to Skype features coupled with the capabilities of Asterisk. For example, the beta version of Skype For Asterisk will allow customers to make, receive and transfer Skype calls from within Asterisk systems using their existing hardware; enable inbound calling solutions like free click-to-call from company websites or virtual offices; and manage Skype calls using Asterisk applications such as call routing, conferencing, phone menus and voicemail.

Hey, I guess I was right in my (Astricon) prognostications earlier today about it having to do with Skype.

The Skype For Asterisk Beta program begins today. Asterisk users, system administrators and developers are invited to apply to participate at http://www.astricon.net/skype

I'm trying to figure out how you transfer a call to a Skype username (i.e. tkeating) using a traditional (Asterisk) IP phone with no keyboard - just a numeric keypad.





Digium AEX410 Launches

September 25, 2008

Digium announced the immediate availability of the AEX410, a four-port modular analog PCI-Express x1 telephony interface card for use with Asterisk. The AEX410 is a PCI-Express board that compliments Digium's existing PCI-based TDM410 product.

The AEX410 offers analog (FXS) stations and analog trunk (FXO) modules for connecting to the PSTN or analog devices. An optional DSP-based 128ms line echo cancellation for the AEX410 is provided by Digium's VPMADT032 G.168 module. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces.

According to the Digium blog, the naming convention is as follows:
AEX4XYZ

Where X indicates the number of FXS (station) modules (ports)
Where Y indicates the number of FXO (trunk) modules (ports)
Where Z indicates either B for bundles not containing DSP-based echo cancellation or E for bundles that do contain DSP-based echo cancellation.

So for example, here are some sample models, though not limited to just these:
AEX422E <- 2 FXS, 2 FXO, has DSP echo cancellation
AEX440E <- 4 FXS, 0 FXO, has DSP echo cancellation
AEX404E <- 0 FXS, 4 FXO, has DSP echo cancellation

The AEX410 board product utilizes Digium's wctdm24xxp driver file that is part of the Zaptel (soon-to-be DAHDI) driver package.















Digium Major Announcement - what can it be?

September 25, 2008

Today, Digium, creator and primary developer of Asterisk, the leading open source telephony platform will be making a major announcement at Astricon later today. Digium hinted to me that a major announcement would be made at Astricon when I visited their Huntsville, Alabama headquarters in August.

I tried to find out what the news will be, but alas Digium couldn't tell me. So I thought it would be fun to prognosticate what this deal could be.

1) Digium's Switchvox will be distributed by Dell, which currently carries another Asterisk competitor, Fonality.

2) Digium will be acquired by Adtran, an avid supporter of Digium in the past.

3) HP seeing that competitor Dell is offering IP-PBXs (i.e.







HUD 3.0 for trixbox CE and any other Asterisk IP-PBX!

September 23, 2008


I spoke with Kerry Garrison last week at ITEXPO and he gave me a news scoop that Fonality would soon offer the HUD 3.0 Unified Communications client for the open source trixbox CE Asterisk-based platform. trixbox CE is one of the most popular Asterisk distributions. I recently commented in my trixbox Pro review (paid version),"The feature-rich HUD Pro client is certainly a competitive advantage Fonality has over many other Asterisk-based solutions." As such, offering HUD 3.0 for trixbox CE is a major move by Fonality.

In fact, Kerry told me that once HUD has been ported to trixbox CE, the new HUD will work not only on trixbox CE but should also work other Asterisk flavors, i.e. Switchvox, PBX in a Flash, Voiceroute, etc.


Fonality Targets Call Centers with Advanced Call Center Features

September 18, 2008


I met with Fonality CEO Chris Lyman at ITEXPO and he gave me a demo of HUD 3.0, which includes some very advanced call center features. HUD 3.0 now not only displays the queues, but it lets you drag-and-drop individual queues off the main HUD client onto your Desktop allowing managers & agents to focus on specific queues of interest. The new version features important statistics such as abandonment rate, ASA (average speed of answer), and more. You can see all of your agents in a particular queue and they are color coded to indicate their status (on internal call, on queue call, etc.)

One critical feature is that if a call is not being answered, it immediately broadcasts a toast popup window to all the agents in the queue and allows an agent to take the call before it is abandoned.


ITEXPO West 2008 a Resounding Success

September 18, 2008

Some great news from TMC about our IP communications conference & expositions (ITEXPO) taking place this week that I thought I'd share. The show is still going on, with today being the last day. Yesterday's exhibit hall attendance was tremendous as seen by some photos I snapped and posted yesterday. I had some great meetings or saw many important VoIP companies, including Asterisk/Digium, CosmoCom, Fonality, Microsoft, Packet8, PIKA, Skype, and more.

HUD3 Launches

September 17, 2008


Apparently, Fonality has decided to launch HUD3 (also called HUD 3.0), the latest version of their latest communications client at ITEXPO. I was aware HUD 3.0 was coming soon, but did not know it was launching at ITEXPO until I saw T-Shirts being worn by fellow TMC team members in the registration area with HUD3 written on them. I used my top secret security clearance badge   to get into the off-limits exhibits area, which was still in the setup phase. I went over to the Fonality booth to see if I could confirm if HUD3 was launching and sure enough their booth display announces HUD3.

I guess Chris Lyman decided ITEXPO was the best place to announce the news.


PIKA WARP Appliance adopted by Schmooze

September 16, 2008

Today, as I'm blogging from the ITEXPO press room I learned that the PIKA WARP Appliance has been adopted by Schmooze (aka the Yiddish Asterisk ). If you recall, I reviewed the PIKA WARP Appliance last week. The WARP Appliance can run Asterisk, FreeSWITCH, and now Schmooze.


Schmooze Communications will be using PIKA WARP the Appliance as the hardware component in a new line of its PBXact business telephone systems.

Schmooze president Tony Lewis and his development team are major contributors to FreePBX, one of the top graphical user interface (GUI) applications in use today that provides pre-programmed functionality for users wanting to ease the configuration of the Asterisk open-source platform. While there are a number of GUIs on the market, FreePBX has emerged as one of the industry standards that, when combined with PIKA WARP the Appliance, provides developers with a superior solution for cost-effective, scaleable and customizable business phone systems.




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