Source: www.voip-news.com

Point. Click. Go. That’s one of those easy, fun things about web-based VoIP solutions. But hosted IP PBX solutions traditionally lacked that capability … until now.

A user dashboard for hosted IP PBX applications? Oh yes. Vocalocity has just released on, bringing click-to-call functionality and on-screen ease to the hosted IP PBX environment, which has traditionally lacked those two things.

According to a recent VoIP-News feature:

The application, called the VocalocityPBX Dashboard, works through a Web browser rather than through separate PC software. It’s available free to all users of the hosted service, allowing them to view onscreen information about the calls, conferences and queues operating in their companies at the moment. It also lets them see the status and availability of co-workers, and set their own status for co-workers to see. Permissions set by management determine exactly who can see what. CEOs, for example, typically can view more information about calls, such as whom each person is talking to, than can other employees.

Source: www.voip-news.com

There’s been some huge raises in IP PBX in the UK market. IP PBX desktop extensions in the United Kingdom  rose 40 percent in the fourth quarter of 2007, according to a study by MZA. Nearly 750,000 PBX extensions were installed in that time frame.

Cost remains an impediment  for small businesses though.

Read about it here.

Source: snapvoip.blogspot.com

Computer world Malaysia is reporting t QubeConnect Sdn Bhd, a Malaysian company has produced to the market it’s own VoIP IP Telephony solution based on open standards. The call control solution, called the QubeTalk Enterprise Communications Server (ECS), unifies voice, data and video on a single IP network and is targeted at medium and small-sized enterprises.
According to the company, QubeTalk ECS architecture also integrates both IP and legacy TDM (Time Division Multiplexing) connections that ensures high availability and minimal downtime as well as full SIP (Session Initiation Protocol) support.
Complete article at CW Malaysia.

Source: www.voip-news.com

3CX has released a new version of their 3CX Phone System for Windows.

The new version of the IP PBX, version 5.1, includes 3CX Tunnel, which channels VoIP traffic over one port. This both simplifies the firewall configuration and makes it easier for remote access to the company PBX.

“We know how important it is for businesses nowadays to have remote workers and branches seamlessly integrated to their company’s phone system, and the addition of the 3CX Tunnel to 3CX Phone System for Windows facilitates this process,” said Nick Galea, CEO. “The 3CX Tunnel, unlike other similar tunneling protocols, is not proprietary and can be used with popular VoIP softphones and hard phones. This is ideal for businesses, as they can use a variety of telephone options with their IP PBX depending on their needs and budget.”

There are four editions of the IP PBX from 3CX – small business, pro, enterprise and free.

Source: snapvoip.blogspot.com

"The Future of Open Source VoIP is Asterisk 1.6" was mentioned a few times in the past here and elsewhere and now it has arrived, at least the beta, Asterisk 1.6 Beta1. I have been plying with it from SVN but the Asterisk dev team announced that the official release of beta to get more people involved in testing.
I idea is more people test the release 1.6, sooner it can leave testing stages and go into main stream release.
Also remember that Asterisk 1.6 will be the first major release of Asterisk since 1.4, which was released just over one year ago. This release contains a number of new features, as well as architectural improvements for improved performance. The changes since asterisk 1.4 could be found here. The Readme file for the release is here.
I also have been testing A New Channel Driver For Asterisk, chan_unistim. with Nortel phones as I have them in a VoIP Solution that I have proposed to an institution and myself is using one of them.
Release info at Asterisk.org

Source: snapvoip.blogspot.com

The Asterisk development team has released Asterisk version 1.4.17 which fixes SIP security issue, as well as a number of other bug fixes.

The SIP security issue is documented in the published security advisory, AST-2008-001. This issue only affects Asterisk 1.4. Asterisk 1.2 is not affected. Systems that do not use chan_sip are also not affected.

The security advisory is here in PDF format.

The release 1.4.17 is available for immediate download.

Source: snapvoip.blogspot.com

The SIP based communications are becoming a preferred interconnection in this growing market, touted by some to be $22 Billion. But as with any communication method, there are some gotcha’s hiding and waiting to pounce on you the day you go production. So how do you avoid the hassle, you test with proper testing procedures and tools. So TekVision is in the market, to take it’s portion of the $22 Billion by providing a testing service, a testing Lab.
TMCNet’s Erik Linask writes how the whole procedure works and how you stand to gain by testing with them.

Source: snapvoip.blogspot.com

The SPA962 VoIP telephone will appeal to businesses using a hosted IP telephony service, an IP PBX, or a large scale IP Centrex deployment, with it’s array of features.

Here are some features that stand out;

• Full featured six line business class IP Phone supporting Power over Ethernet 802.3af
• Connect directly to an Internet Telephone Service Provider or connect to an IP PBX
• Dual switched Ethernet ports, Speakerphone, Caller ID, Call Hold, Conferencing, and more
• Appealing Four Inch, True Color Liquid Crystal Display (LCD)

I like the fact that it has a color Screen!Standard features on the SPA962 include six active lines, dual switched Ethernet ports, 802.3af PoE support, a high resolution color display, speakerphone, and a 2.5 mm head-set port. Each line can be independently configured to use a unique phone number (or extension), or can be configured to use a shared number that is assigned to multiple phones. The power supply for the SPA962 is sold separately and will be required if PoE functionality is not implemented.

Comprehensive Interoperability and SIP Based Feature Set
Based on the SIP standard, the SPA962 has been tested to ensure comprehensive interoperability with equipment from VoIP infrastructure leaders enabling service providers to quickly roll-out competitive, feature rich services to their customers. With hundreds of features and configurable service parameters, the SPA962 addresses the requirements of traditional business users while leveraging the advantages of IP telephony. Features such as easy station moves, presence, and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA962.

Carrier-Grade Security, Provisioning, and Management
The SPA962 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Linksys secure remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high quality support to their subscribers. Remote provisioning also saves service providers the hassle and expense of managing, pre-loading, and re-configuring customer premise equipment
linksys

Source: snapvoip.blogspot.com

FCC, Federal Communications Commission, states that "Voice over Internet Protocol (VoIP), is a technology that allows you to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line. Some VoIP services may only allow you to call other people using the same service, but others may allow you to call anyone who has a telephone number – including local, long distance, mobile, and international numbers. Also, while some VoIP services only work over your computer or a special VoIP phone, other services allow you to use a traditional phone connected to a VoIP adapter."
Let’s see how others describe it.
Howstuffworks tries to give you much information as possible with the article "How VoIP Works". I have gone through the complete article and found it to be ok. My writing ability is not as good as Robert Valdesz (It is getting there), otherwise I would try to write a simpler version as well.
He starts with how VoIP Works, How current PSTN, Public Switched Telephone Network, works which continues to the bits and bytes involved in a regular Telephone call, Advantages of VoIP, Disadvantages of VoIP and Codecs, softswitches and protocols.

All in all you get an idea of How VoIP Works but I wonder if the article ib being kept updated, for instance about E911, A lot has changed since it was written. So that is "How VoIP Works" until my writing improves. If I find any other VoIP introductions, I will Post them here.

Source: snapvoip.blogspot.com

After acquiring SwitchVox, Digium has released a free version of the SwitchVox. This was announced at the Fall VON, but the press release was buried under all those notes I took at Fall VON. Better later than never and here is the Press Release.
Also you can download the release directly from SwitchVox site. It is great way to test or have a trial run of the software which comes in a easily installable package. But for production, it is suggested that the the commercial version, SOHO or SMB, is used. The package includes, Fedora Core 6 Linux, the open source Asterisk PBX software, plus all the software necessary to make Switchvox such a revolutionary communications solution.

BOSTON (Fall VON and Digium|Asterisk World) — October 31, 2007 — One month after its acquisition of Switchvox, a leading provider of IP PBX phone systems for small- and medium-sized businesses (SMBs), Digium®, Inc., the Asterisk® Company, announced the immediate availability of Switchvox Free Edition. The new product includes the telephony features of the widely used Switchvox SOHO edition and can be downloaded and installed in minutes on existing hardware. The release underscores Digium’s commitment to delivering full-featured telephony systems at low price points to allow organizations around the world to benefit from the Asterisk open source movement and flexibility of voice over IP (VoIP).

Switchvox products are based on Asterisk®, the open source telephony software created and owned by Digium. Switchvox Free Edition includes tools such as an interactive voice response (IVR) editor that allows system administrators to create auto attendants and menus. It also includes features such as voicemail to email, find me/follow me, unlimited calling queues (automatic call distribution, or ACD) and advanced reports on system use.

Switchvox Free Edition supports Digium’s line of analog cards. Users can install Switchvox Free Edition on existing hardware or use Switchvox-certified hardware. For details on hardware support, see www.switchvox.com/certified_hardware.

“Asterisk is the most widely deployed open source telephony platform, and by introducing Switchvox Free Edition, Digium has the opportunity to introduce the technology to a broader and more mainstream customer base,” said Tristan Degenhardt, Switchvox product line director at Digium. “We anticipate thousands of downloads in the coming weeks by companies interested in a better way to communicate with their customers. The smallest businesses will find Switchvox Free Edition to be an excellent way to get started with a full-featured phone system and be positioned to easily transition to more advanced Switchvox systems as their needs change and grow.”

Switchvox Free Edition is on display this week at Fall VON 2007 and Digium|Asterisk World at the Boston Convention and Exhibition Center. The product may be downloaded from www.switchvox.com

Source: snapvoip.blogspot.com

The Appliance for Linux by Pika Technologies is the second member of Pika’s Warp family, following last month’s release of Appliance for Asterisk,
Pika Technologies has introduced a compact, customizable, Linux-based IP-PBX appliance. Based on the company’s host media processing software, "Pika Warp, the Appliance for Linux," is said to provide IP-PBX (IP-based private branch exchange) functionality along with integrated voice response (IVR), predictive dialing, and appointment reminders.
Announced features and specs of the Appliance for Linux include:

Processor — AMCC Power PC 440EP, clocked at 533MHz (upgradeable to 666MHz)Memory — 256MB RAM; 64MB flash (on SD card)Display — 2 x 20 backlit LCD display, with API-controlled front-panel scroll buttonEthernet and USB ports (number and type unspecified)Music-on-hold audio in; paging system audio outIP and analog telephony I/O:up to 100 IP trunk/station interfacesone standard FXS (foreign exchange subscriber) interfaceup to eight additional FXO/FXS (foreign exchange office/subscriber) interfacesPower failure transfer RJ11 jack(s)Dynamic thermal managementOperating system — Denx ELDK, with a 2.6.19.2 Linux kernel
Linux Devices has complete detail.

Source: snapvoip.blogspot.com

TORONTO, ON/SEOUL, KOREA – October 17, 2007 – Sigma Systems (www.sigma-systems.com), a premier provider and leader in the design, development and deployment of OSS service management solutionsand Xener Systems, Inc., (www.xener.com), a key provider of Next Generation Networks, of Seoul, Korea, have signed a strategic alliance to accelerate and collaborate on voice over IP (VoIP) solutions in Korea and Asia Pacific.

“Softswitch vendors now recognize the value of an integrated service management platform to enable operators to deliver an end-to-end solution as an integral part of a complete service delivery model,”says Sigma Systems president and COO Tim Spencer. “As such, we have forged a strategic alliance to embark on a joint marketing and sales strategy that will provide a holistic solution for Xener’s current and prospective customers in Korea and APAC.”

“With this agreement, we gain momentum to expand into new markets by taking Sigma Systems’ residential and commercial VoIP expertise and offering a value-added solution that addresses the voice needs of cable and telecommunications companies,” adds Xener Systems CEO Yong-Gu Kang. “Through our mutually beneficial business collaboration, we will leverage our own next generation network capabilities and add Sigma’s service management solutions to expand our voice services business. We will position Sigma as a valued partner as operators expand or transition services for competitive or operations scalability opportunities.”

The Sigma VoIP Service Packages provide a robust OSS service management solution that is pre-integrated to define, provision and maintain residential VoIP and hosted commercial VoIP services on IP networks. The service package encapsulates the voice services domain expertise and best practices that Sigma has gained from working closely with some of the world’s largest communication service providers since 1996. The solution is applicable to any cable, telco or broadband IP service provider looking to offer residential and/or business voice services.

“Our solution will allow quick optimization and automation of operational processes for voice services for many of Xener’s customers,” adds Spencer. “It can rapidly create new bundles of residential and commercial services, including hosted voice service for small to medium businesses (SMBs), voice mail, unified messaging, SIP residential telephony features and Centrex and IP-Centrex call features for SMBs.”

“As the VoIP service market is quickly growing in Korea and Asia Pacific, Xener has focused its efforts in enhancing responsiveness to the customers’ needs on the diverse and differentiated subscriber services by expanding its partnerships with the industry’s best-of-breed solution providers like Sigma Systems,” adds Yong-Gu Kang. Xener has maintained its leading position in NGN and VoIP solution market, deploying the core networks for most of VoIP service providers in the Korean VoIP market such as KT, Hanaro Telecom and Korea Cable Telecom (KCT), and recently extended its coverage into the enterprise IP telephony solution market with its own IP PBX and IP-Centrex solution.