IPChitChat Blog

3CX version 7.1 has now completed beta testing and will be released later this month for download. The following key enhancements are available in version 7

3CX Assistant - This application is installed on the desktop and allows users complete call control, as well as extensive presence and queue information. The 3CX assistant works in tandem with a software or hardware IP phone and gives users the ability to pick up, divert, transfer and park calls with a mouse click. Management level users can monitor all users, groups and calls, dragging and dropping of calls to users is also supported!

Inbuilt reporter - the separate call reporting utility has been replaced by an inbuilt call reporter with extensive reports, including new Call Queueing reports, it's a big improvement over previous reporting tools and is well worth a look.

BCDR service - A separate service can now output CDR (Call Data Records) of calls in real time. It can output to a text file or to TCP (Hostname and port) and the format can be configured. CDR is a widely used format used for billing, particularly in the hospitality sector.

Grandstream has announced that release 1.0.1.21 for GXW4004/GXW4008/ GXW4024/ HT502 is now officially released. The firmware and release notes can be downloaded from:

http://www.grandstream.com/firmware.html (Please refer to the related product on the page for firmware download or release note ).

Digium has announced the availability of version 1.5 of their AsteriskNow software appliance product. The most interesting addition is the introduction of FreePBX GUI as a replacement for the version 1 AsteriskNow GUI. The download is available from the AsteriskNow site:

http://www.asterisknow.org/downloads

Snom today announced its new sleek looking 820 Business phone.  The 820 is the first in a new series of business phone from Snom and features a high resolution display in addition to POE and WiFi.

There is more information here:

http://www.snom.com/en/products/snom-820/

 Anyone interested in Skype for Asterisk?

Skype For Asterisk beta program starts today, adding Skype features to Asterisk-based solutions

GLENDALE, Ariz. - Digium(R), creator and primary developer of Asterisk(R), the leading open source telephony platform, and Skype(TM), the leading global Internet communications company, today announced the beta version of Skype For Asterisk, 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.

"Throughout our individual histories, Skype and Asterisk have each disrupted conventional communication methods through innovative, cost-effective solutions," said Stefan Oberg, vice president and general manager for Skype Telecom and Skype for Business. "We are excited to be working together with Digium to offer small and mid-sized businesses an even more powerful communications solution to conduct business worldwide."

Specifically, the beta version of Skype For Asterisk is an add-on channel driver module that integrates Skype Internet calling with Asterisk-based telephony products. Skype For Asterisk also complements small and mid-sized business users' existing services by providing low rates for calling landline and mobile phones around the world.

"Working together with Skype, our goal is to help businesses boost productivity and reap the rewards of feature-rich telephony software, all while saving a substantial amount of money," said Danny Windham, CEO of Digium, the creator and sponsor of Asterisk. "The Skype For Asterisk beta program is a first step towards adding Skype capabilities to Asterisk-based phone systems and enabling them to reach more than 338 million Skype users."

The beta version of Skype For Asterisk will enable business users to:

• Make, receive and transfer Skype calls from within Asterisk phone systems, using existing hardware.
• 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.

Following the beta period when the product is released, Skype For Asterisk will be sold and distributed by Digium and its worldwide network of resellers.

Live at AstriCon

Stefan Oberg will provide the first public demonstration of Skype For Asterisk during his keynote address today at AstriCon, the annual Asterisk user and developer conference. AstriCon attendees are also invited to stop in and see a demonstration of Skype For Asterisk at the Skype booth on the expo floor.

Skype For Asterisk Beta Program

The Skype For Asterisk beta program begins today; Asterisk users, system administrators and developers are invited to apply at http://www.astricon.net/skype. The initial beta is limited to a select number of users, developers and integrators.

Grandstream are holding a webcast for those of you interested in overcoming NAT issues with the GXE series PBX appliances. Details are as follows:

Wednesday, October 1, 2008 10:00 AM - 11:00 AM EDT

https://www1.gotomeeting.com/register/260927172

• Join Grandstream for a discussion on how to mitigate NAT traversal issues when deploying the GXE502x IP PBX. Topics covered will include:
• How to deal with the NAT traversal when deploying the GXE502x behind another router
• How to support remote teleworkers working from behind home routers over the Internet
• How to support peer-office GXE502x IP PBX systems that are deployed behind routers
• -What is UPnP, how it works with the GXE502x and what are the limitations
• What major SMB routers on the market have been successfully against the GXE502x on UPnP
• What is STUN, how it works on GXE502x and when we consider using it.
• Other tactics in dealing with NAT traversals

It is quite unusual to hear of a large organisation taking the plunge and moving from an established platform such as Cisco Callmanager into the world of open source. That's why I was quite interested in the following article:

http://www.networkworld.com/news/2006/091206-von-sam-houston.html

It details how Sam Houston State University are in the process of replacing their Cisco CallManagers and legacy Meridian PBXs with an Asterisk based phone system.