TrixBox Alternatives.

http://snapvoip.blogspot.com/

Fine product as it was, Trixbox CE was killed by Fonality in 2012. If you are still clinging on to TrixBox, there are plenty of alternatives that are even better and keep you secure through timely updates and support,
Here are some of those (the list is organic and we will update is as necessory);

FreePBX GUI and Distro

The FreePBX project has been around for a very long time and we have written a whole lot about FreePBX. FreePBX is one of the most widely used management tool for Asterisk deployments. But it has much more to offer than just a management tool, the distro ISO comes with all the tools you need to run a super duper telephony system. The Distro is based on CentOS and you can select either 32bit or 64bit befitting your needs. Download from here.

AsteriskNOW IP PBX

AsteriskNOW is another fast and easy way to  way to get started Asterisk installed and start building custom telephony solutions or simply use as a telephony solution. We have been fans of AsteriskNow for a long while. Like FreePBX, you will have to download the .iso file, burn it to a CD, drop it into the CD or DVD drive on the target computer to build a complete Asterisk telephony solution. The installation will provide you with,  complete Linux distribution, CentOS again, Asterisk, the DAHDi driver framework, an administrative GUI (Thanks to FreePBX project above), the MySQL database, the Apache web server and a wide variety of development tools and components. Just like FreePBX, you can download 32bit or 64bit versions of AsteriskNOW.

Elastix

Elastix is a fine open source VoIP IP Telephony software package which brings together best-in-class tools for the task into a single, easy-to-use interface. Elastix a prominent IP Telephony distribution that is growing everyday. It has been translated into dozens of languages, and carries extensible modules and advanced features. Elastix is available to download from Elastix download site and it is also available for Raspberry PI with the moniker uElastix.
 

PBX in a Flash

Yes it used to mean on a flash drive but I am happy to say now I am referring 'FLASH' to the speed, in implementing a telephony solution based on Asterisk. An amazingly versatile turnkey PBX is based on the latest versions of popular Linux distribution, CentOS and your choice of Asterisk 1.4 or 1.6 with FreePBX, a Linux firewall, and Fail2Ban provides enhanced security.

SIPExchange 1.1.0b From CafeSIP Is Ready For Download

SIPExchange 1.1.0b snapvoip.blogspot.com
SipExchange is a project by CafeSIP and we have been following the CafeSIP projects for a while. In addition to SipExchange, they also produce;

I is a framework for developing SIP server-side applications using the Java programming language.
SipUnit is a unit testing tool for SIP server-side applications and a bunch of common projects.

SipExchange, a feature-rich SIP softswitch, (a open-source softswitch) that provides standard SIP services like location, proxy and presencethat can be used by service providers to offer voice, video, chat and other multi-media services to their subscribers. SipExchange could be used in many different ways to manage your communication needs;

1. Service providers can use SipExchange to offer SIP-based voice, SIP-based video and data services to their customers.
2. Internet service providers can host specialized SIP services for their corporate customers using SipExchange.
   
3. SipExchange can be used as an IP PBX for communications within an office or even inside a home.
4. Companies can provide specialized SIP-based services to their customers and partners by extending SipExchange.
5. Software vendors can offer specialized SIP-based services to their customers by extending SipExchange.

 You could download SipExchange and other products from CafeSIP.org

SoftSyl Selects Broadvox' GO! SIP trunking..

SIP Trunking snapvoip.blogspot.com/
Press release from Broadvox announced that Softsyl has selected Broadvox' GO! SIP Trunking which will be incorporated into Softsyl gateway. SoftSyl will also provide GO! SIP Trunking to existing customers as well as providing an IP IVR solutions incorporating GO! SIP Trunking with SoftSyl UVX Gateway.
Here is the press release

February 24, 2010 - Broadvox, a leading provider of IP Communication solutions for SMBs and enterprises, announced today that SoftSyl Technologies has selected GO! SIP Trunking to be offered with their gateway as a complete product. This enables SoftSyl to provide cost-effective, all-in-one IP IVR solutions incorporating GO! SIP Trunking to businesses that purchase a SoftSyl UVX Gateway or to offer GO! SIP Trunking itself to existing customers.

Broadvox has been a leader in the IP communications marketplace for years, delivering customized SIP Trunking solutions with built in reliability, scalability and interoperability. SIP Trunking offers numerous opportunities to increase profits with unlimited local calling and lowered long distance rates while using IP PBXs or IADs. SMBs realize an almost immediate ROI as a result of the reduction in cost per minute for communication provided by SIP Trunking and save as much as 70% when compared with traditional TDM.

"We are pleased to offer Broadvox products in our customized communication solutions to both new and existing customers,” stated S. Jalali, Vice President of SoftSyl Technologies. Broadvox is recognized for their reliable, carrier-grade voice quality and when packaged with the SoftSyl UVX Gateway, our customers can achieve all the features and functionality of GO! SIP Trunking while enjoying even bigger savings in telecom costs with our unified package as they migrate to IP communications or upgrade their existing infrastructures."

"An extensive list of components are combined in the SoftSyl UVX Gateway that dramatically reduce the cost of IVR and VoiceXML powered solutions,” said David Byrd, VP of Marketing and Sales at Broadvox. “When Broadvox SIP Trunking is deployed in combination with the SoftSyl UVX Gateway, it results in significant cost savings when compared to communication systems that require separate phone and data connections. This is an opportunity for increased profitability, exposure and business growth, for both Broadvox and SoftSyl." 

Broadvox Press Release

Setting Up An SIP Extension In Elastix (Video)

Elastix How to snapvoip.blogspot.com

Adding extensions makes one of the first task of setting up an IP PBX.  In this video you can see how to add one of the basic extensions, a SIP extension to Elastix. It is presented in very clear and to the point by Scott Wittenberg. This method is the way you add any simple extensions to Elastix. We will be posting more videos from the Elastix Youtube Channel. in the future.

snom And Cudatel Partner To Provide Best Of OSS IP Telephony

snom partners with Cudatel snapvoip.blogspot.com

snom making partnerships with eqipment makers to provide end to end VoIP IP Telephony solutions to small to medium sized business'. The latest is their partnership with Cudatel,  a wholly owned subsidiary of content security vendor Barracuda Networks Inc. snom is know world over for developing and manufacturing voice over IP (VoIP) phones for enterprise and residential markets.

The partnership paves the way for businesses to deploy snom's portfolio of Linux-based SIP VoIP phones with CudaTel's FreeSwitch based IP PBX and communications appliance, CudaTel Communication Server. FreeSWITCH is a free open source software communications platform for voice and chat-driven products released under the MPL free software license.
snom and Cudatel together have conducted rigorous interoperability and certification testing to ensure the joint solution is fully integrated and deployment ready and that the two companies' gear will integrate seamlessly and ease VoIP deployments.

We are pleased to be working with CudaTel as we continue to partner with the best-in-class technology partners, CudaTel is taking the expertise Barracuda Networks established with its data communications infrastructure product portfolio and bringing this same innovation into open source VoIP with a robust, FreeSWITCH-based platform. Teaming up with CudaTel creates a full, end-to-end VoIP solution for companies of any size built on an open source model, said Dr. Michael Knieling, Executive Vice President of Marketing and Sales for snom Technology AG.

 The CudaTel Communication Server is offered in four models with prices starting at $2,000. The IP PBX are capable of supporting 25 to thousands of VoIP telephones with a range of advanced services. These advanced services include automated handset provisioning and deployment call routing, call recording, call queuing, parking, unified messaging and voice mail, conferencing and conference bridges. The  CudaTel Communication Server is also capable of providing automated attendant services, Active Directory integration and follow-me functionality. CudaTel also offers unlimited 24x7 phone and email support with each CudaTel server and does not require per-user, per-box or per- feature licensing fees.

 CudaTel was built from the ground up to be an innovative and affordable VoIP system and designed to work exclusively with best-of-breed SIP phones like the endpoints offered by snom, Partnering with snom ensures that our CudaTel Communication Server customers will have access to telephones that are easily deployed and provisioned, and that offer a host of capabilities and functions. said Sean Heiney, CudaTel product manager and director of new product initiatives for Barracuda Networks.

 With the new partnership, expect to use  snom's portfolio of SIP-based commercial-grade VoIP phones, including the snom 320, snom 360 and snom 380, as well as the new snom 820 and the snom 870 color touch-screen phone with Cudatel eployments, existing or planned. Also supported are the snom MeetingPoint conferencing phone and the snom m3 IP DECT phone from snom. snom was the first company to provide "zero touch" IP Phone deployment by following the TR-069 standard.
snom Technology AG.
Cudatel

Digium / SwitchVox Gets A New Nothern California Partner, Teledynamic Communications Inc

Switchvox IP PBX
Digium, the Open Source Asterisk base Switchvox IP PBX creator has entered in to an agreement to have Teledynamic Communications Inc as a representative in Northern California. With the agreement, Teledynamic will be able to provide San Francisco bay area business with access to economically priced, open source based IP PBX, SwitchVox.

"Businesses are looking to squeeze more out of smaller IT budgets while still driving to improve productivity through smart uses of technology. The Switchvox provides Unified Communications, Google and Salesforce integration, affordable conferencing and much more for up to 50% less than the proprietary manufacturers. Businesses just aren't willing to pay a premium in this new world we live in." says Randy Kremlacek, President of Teledynamic Communications.

Digium is the creator of Asterisk, the open source voice communications software deployed by more than 3.5 million servers worldwide. Digium Asterisk (Switchvox) is the most widely used Open Source PBX, and dominates in market share with more than 85% of the Open Source market, with Aastra and Fonality the next two largest players.

Teledynamic Communications was founded in 1979 and serves the greater San Francisco, Oakland, and San Jose area, providing consultation, pbx installation, maintenance, remote monitoring and support plans. The company specializes in call centers, audio and video conferencing, unified communications, hosted PBX and VOIP trunking.

Teledynamic Communications

Digium Switchvox

Tags: Unified Communications, Asterisk, digium, Switchvox, Teledynamic, Open Source PBX, IP PBX

Microsoft Response Point IP-PBX Dies.

Microsoft Kills Response Point IP-PBX
Tom Keating is telling us that Microsoft has just killed the Response Point IP-PBX, which he says was a pricey but decent product. I myself did not use it but during various visits to Redmond, I was shown the product, because I was interested in UC, Unified Communications.
The IP-PBX seem to be a good challenger from outset but I liked the Microsoft Office Communications Server, OCS 2007 much better in the way of Unified Communications. I have a few pals who live by it. But it is mainly for Enterprise level deployments and I wonder what Microsoft has for the SMB Market? May be Small Business Server with Microsoft OCS component?
Andy Abramson has a point, except for a XBOX, I have not seen many Microsoft hardware that flourished.
Tom Keating via Andy Abramson

Tags: IP PBX, Microsoft, OCS 2007, Response Point IP-PBX, Office Communications Server

SIP Softphone Application for the iPhone and iPod Touch, Media5-Fone.

Media5-Fone
Media5-fone from Media5, provides iPhone and iPod users with ability to connect with any SIP IP-PBX, be it an Enterprise telephony service or Service Provider Network, like the ones we have listed in The Free SIP Account Provider List
Media5-Fone lists following features;

Pure SIP standard softphone	Voice Mail Integration	Loudspeaker
VoIP over Wi-Fi	Native Contacts List	Hold
Easy Configuration	Call History	Mute

According to the Media5 press release, "the Media5-Fone enables offices as well as the mobile worker to use their iPhone as an IP-PBX extension, allowing them to access the same phone services and features as if they were in the office. Remote workers can be in contact with any other office or employee easily and efficiently, promoting collaboration across all locations.
VoIP service providers who offer calling plan can also benefit from the same SIP connectivity extension for their customers who own an iPhone."

"This new release of the Media5-Fone extends our mobile portfolio to iPhone users on the go, and offers them the key features needed to integrate an easy-to-use SIP IP-PBX extension within the iPhone,” explained Pascal Doré, Mobility Product Line Manager. "In addition to this Lite version, our engineers are already at work to bring the next fully featured Enterprise version of the Media5-Fone that will embed strong Voice security encryption among the key features. Worth mentioning that the Media5-Fone can be co-branded and custom-tailored for any specifics system needs."

The Media5-Fone application for the iPhone is available in the Apple iTunes.

Media5 Press Release could be found here.

Tags: Media5-Fone, Free SIP, SIP, iPhone, iPOD, VoIP over Wi-Fi, SIP connectivity, IP-PBX, Free SIP Account Provider List

Polycom VVX 1500, Business Video Phone

http://snapvoip.blogspot.com/
Polycom has released a new video phone VVX 1500, a business media phone that combines a personal video conferencing system with a fully featured voice over IP (VoIP) telephone with Polycom HD Voice(TM) and an open application programming interface (API) and microbrowser for real-time delivery of personalized Web content as well as integration with key business applications.
"There is growing demand from our service providers and customers to help them configure video within our BroadWorks call control platform," said Mike Tessler, CEO of BroadSoft. "We have a long history of teaming with Polycom to deliver high quality hosted VoIP solutions, and the VVX 1500 is especially compelling because it goes far beyond the functionality of a traditional video phone by combining rich telephony, business-class video and an applications platform that is all deeply integrated with the BroadWorks platform, and it is extremely easy to use." Broadsof has been working closely with Polycom bringing the VVX 1500 to the market.
With such advancements in video technology, Polycom is yet to learn technologies like YouTube or vimeo. Otherwise I could have embeded a demo video (WMV only) available at Polycom.
According to Polycoms site;

Features and Benefits

• Intuitive, color touch-screen interface for voice, video and applications
• Six-line, feature-rich phone with Polycom HD Voice™
• Instant, one-touch business-grade video conferencing right from the desktop
• Adjustable camera, base, and display to suit the environment and provide eye-level visual interactions
• Highly customizable applications platform with open Polycom XML API , integrated microbrowser and USB 2.0 for applications
• Bundled with productivity and personalization applications, including Polycom Productivity Suite, Polycom My Info Portal and Digital Photo Frame
• Deep IP PBX integration with a variety of Polycom Voice Interoperability Partners (VIP)
• Integrated Gigabit Ethernet (GigE) switch for bandwidth-intensive applications
• IEEE 802.3af powered at under 11W maximum for a "green" world
• Smart motion detection to enable the screen to go to power-save mode when no one is in the office

More info from Polycom product page.

Tags: video phone VVX 1500, video phone, Polycom HD Voice, BroadWorks platform, voice over IP, VoIP, IP PBX, GigE, IEEE 802.3af, Green Phone

QueueMetrics 1.5.0 released

http://snapvoip.blogspot.com/
Call center monitoring solution for Asterisk IP PBX provider QueueMetrics has released their "QueueMetrics 1.5.0" earlier this month. Among a plethora of other features and improvements, this version introduces a new component called AGAW (Agent Awareness) that lets your agents see in real-time how they are faring comparing to their queue, and puts in place an IM infrastructure where agent-to-supervisor and agent-to-product-specialist communication can be completely off-band.
According to the news release, this feature is currently experimental and if you'd like to test it you might want to read the section "The Agent Awareness subsystem (AGAW)" on page 72 of the User manual.

Tags: QueueMetrics 1.5.0, QueueMetrics 1.5.0, asterisk, call center, Agent Awareness subsystem, IP PBX

YATE 2.0, Has Been Released.

Yate 2 has been released after going through Alpha and beta cycles testing. But it comes at a time when the Open Source telephony is going through some changes in the SIP frontier at the moment. As we reported OpenSER became Kamailio and got a new fork called OpenSIPS.

YATE 2.0's new features include analog lines, robbed bit, SS7, passive recording on E1/T1 and analogic, clustering, high availability, MGCP, and Jingle. libpri was replaced with ysig. In YateClient, jingle support was added and the interface was improved. The GTK interface was replaced with a Qt interface.

• Support for more hardware interfaces and protocols - added SS7, analogic support, RBS, better ISDN, passive recording
• Clustering, balancing and failover support, Linux-HA integration
• Improved client functionality - switched from GTK to Qt
• Easier involvement of the Yate community - swtched from CVS to SVN
• Added MGCP and Jingle support

Yate for me looks to be next-generation telephony solution. YATE is focused on Voice over Internet Protocol (VoIP) and PSTN, at the moment but it is capable of being easily extended and Voice, video, data and instant messaging can all be unified under Yate's flexible routing engine. Also taking into the concideration that YATE is scalble with clustering which most of my favorite Telephony Engines are still trying to achive. So Keep an eye on Yate. Your calls may travel through this system somewhere on line.
You can download, and learn more about Yate at the YATE Project.

Tags: IP PBX, Telephony Engine, YATE, OpenSER, OpenSIPS, SIP, VoIP Engine

Open Telephone Training Seminar Rescheduled.

FreePBX team has announced a date change on the upcoming Open Telephone Training Seminar. When I published the above article, it was scheduled for August.

The FreePBX team has received numerous requests from possible participants to postpone the date because of vacations and other summer time activities in August. After verifying with those who have already signed up for the event, the team has set the following dates for the training seminar;

Where: Digium Headquarters, Huntsville, AL
When: October 7-10th, 2008

For a limited time, until the End of July, we will also provide an early bird discount of 15%, a $375 savings, but register early to get that discount.

Tags: Open Telephone Training Seminar, OTTS, FreePBX, Digium, IP PBX, PBX, VoIP Training

FreeSwitch Gets A Web Based Management Portal From Wiki PBX.

FreeSwitch, another big player in the open source IP PBX arena has got a web front end for management. I have a copy of FreeSwitch running (actually a dormant VM image) which I will be using to test the PBX Management front end. It is available from Wiki PBX and if you are a FreeSwitch user, I think you will be a happy camper.
According to the site, following are the prominant features!;

• Multiple 'accounts' per server instance - each account is effectively a completely independent PBX. (multi-tenant)
• Layered configuration - XML files goes on top of what is stored in database. Allows you to use a database, but stays out of your way if you choose to use flat files.
• Configure extensions, SIP endpoints and gateways via web interface
• Simple Voicemail system written in Python as an IVR (w/ source code)
• View/hangup/transfer live calls
• View call history (CDR records) over the web interface
• Web interface for managing (add/edit/update) IVR's written in Python, Javascript or Java
• Easily record "sound clips" for use in dialplan or IVR's
• Inject audio or text-to-speech into live calls

Here is the link to wiki pbx.
Of course you can find FreeSwitch here.

Tags: FreeSwitch, Wiki PBX, Voicemail system, open source IVR, PBX Management

ASUS Selects Nortel and Microsoft Solutions For Unified Communications.

Taiwan’s ASUS Embraces Unified Communications with Nortel and Microsoft Solutions

Integrating VoIP, IM, Conferencing into a Seamless Real-Time Experience

MAY 20, 2008 TAIPEI, Taiwan - ASUSteK Computer**, one of the world's largest computer and IT suppliers, is powering new levels of productivity and collaboration for its employees through unified communications solutions from Nortel* and Microsoft.
With this deployment, ASUS's employees are now benefiting from one-click access to VoIP, instant messaging (IM), presence, web conferencing and multimedia applications. They can quickly see a co-worker's presence status on their computers and quickly find the best way to make contact at the click of a button, saving time from repeatedly calling busy numbers or leaving messages when people are out of the office. Collaborations become instant when employees initiate web conferencing without having to wait to meet face-to-face, increasing productivity and collaboration. Nortel and Microsoft's unified communications also centralize ASUS's equipment and integrate its headquarters, branch offices and subsidiaries in Taiwan and China into a seamless communications unit at a realistic cost.

Unified communications is expected to help ASUS increase sales and improve customer satisfaction by using secure IP communications connections, either in the office or off-site, to reduce response times to customer requests. The solution will enable the customer service representatives to find a subject matter expert instantly, regardless of location, and to determine if that representative is available.

Nortel Communication Server 1000 and Microsoft Office Communications Server 2007** are the pillars of ASUS's network today with the flexibility to easily evolve to meet the company's future needs. Resiliency and security features ensure that the network stays up and operational while communications and users are protected. Nortel Contact Center solutions have also been deployed to manage increasing customer service requests, making sure that their needs are promptly met by the best agent available. Interstate Communications is Nortel's partner for this project.

"In a hyperconnected enterprise, there are multiple ways of communication that are not interconnected to each other," Y. K. Tsai, general manager, Nortel Taiwan said. "With Nortel's expertise and experience in delivering unified communications, we ensure customers like ASUS can streamline their business processes by eliminating time-wasting delays from tracking down the right people and information. Nortel has realized over US$12 million in annual savings with a 10-month return on our own internal unified communications investment. Through our alliance with Microsoft, we did what we do best for ASUS - simplifying and transforming their business communications to deliver our promise of Business Made Simple."

Nortel's comprehensive unified communications portfolio provides an end-to-end solution that includes multimedia, voice, video, telephony applications, and a complete data and security infrastructure with various deployment models. An integral part of Nortel's unified communications vision, Nortel Communication Server 1000 is a server-based, full-featured IP PBX that provides the benefits of a fully converged network plus advanced applications and over 750 world-class telephony features. The CS 1000 is also the first Microsoft-certified IP-PBX for Microsoft Office Communication Server 2007.

Nortel and Microsoft recently announced the availability of four new joint unified communications solutions designed to transform how people around the world connect, communicate and collaborate. The joint products include the Converged Office unified communications suite, Multimedia Conferencing, and solutions for branch offices and hosted service providers. Since the formation of the Innovative Communications Alliance in July 2006, customers have purchased and deployed more than 430,000 licenses of the joint technology. Customer adoption continues to accelerate with more than 600 wins for Nortel and Microsoft unified communications products since the formation of the alliance.

Gartner Inc, the world's leading information technology research and advisory company, has positioned both Microsoft and Nortel in the Leaders quadrant in the '2007 Unified Communications Magic Quadrant' report. To view Gartner's Magic Quadrant for Unified Communications, 2007 click here.

Tags: Unified Communications, Nortel, Microsoft, ASUS, IP PBX

Matrix IP PBX From India Uses Utilizes The DSP Chip-level Multimedia Processing Solution From Surf

Even though all telecommunications giants are mostly in USA, Japan and Europe, there are other sources getting together to provide VoIP solutions to the masses. Israel based Surf Communication solutions and India based telecommunications equipment manufacturer, Matrix Telecom have come together to produce one of the best IP PBX solutions. Following is the press release;

Yokne'am, Israel (ANTARA News/PRNewswire-AsiaNet) - The Surf DSP chip is a complete media processing solution offering simultaneous support for multimedia convergence - Voice, Video, Data (Fax/Modem) - all running on a single DSP

Surf Communication Solutions ("Surf"), a leading provider of multimedia embedded processing solutions, announced today that Matrix Telecom Pvt. Limited ("Matrix"), an India-based company that designs, develops, manufactures and deploys telecomsolutions, has selected Surf's DSP chip-level multimedia processing solution, the Surf DSP-24(TM), for inclusion in two new products: a VoIP gateway and an IP PBX solution.

Matrix's feature-rich 32-ports VFX32 VoIP gateway utilizes the Surf DSP-24 to offer IP connectivity with excellent audio quality for analog phones, legacy PBX systems and analog faxes. It is scalable for lower configurations and various FXS and FXO
combinations can be achieved which will help customers to get the lower configuration as per their need.

The Matrix IP PBX offers a complete solution to meet the requirements of voice communication for businesses with multiple locations, traveling employees, and/or home-office networks. The incorporation of Surf's versatile yet powerful DSP-based multimedia processing platform allows the Matrix IP PBX handle high densities for a variety of codecs over converged networks.

The Matrix IP PBX is SIP-based and supports a wide range of telephony applications, such as Voice Mail, Auto Attendant, Call Forward, Call Transfer, Call Park, Call Pick up, Do not Disturb, Hunting Call/Department Call, Time Based Call Routing, Number Based Call Routing/ Least Cost Routing, and Least Cost Routing for multiple ITSP as per defined priority for cost saving.

In addition, the Matrix IP PBX supports RADIUS client protocol which allows the IP PBX to store/log calls in the remote server and provide DDI routing and CLI-based routing.

The Surf DSP-24, based on Texas Instruments' C64x(TM) DSP generation, is a media processing solution offering simultaneous support for multimedia convergence-Voice, Video, and Data (Fax/Modem) -all running on a single DSP.

Equipment manufacturers who develop Media Gateways, IP PBX, CTI products, and other media-over-packet applications find this solution ideal and optimized for their requirements, as well as quick and easy to integrate into their hardware designs. All Surf products support an easy migration path from voice to video, since the same API supports all media types.

"We chose Surf's solution for its ability to support triple-play voice, video and fax/modem on the same platform, which will help us to expand our product range and features in the future," said Mr Ganesh Jivani, CEO, Matrix Telecom. "In addition, the comprehensive documentation set and sample applications were instrumental in achieving fast time-to-market for both products."

"The competition couldn't match the densities or price of the C6424-based Surf platform," stated Ashvin Navadia, Sourcing Manager, Matrix Telecom.

"We are pleased to be selected as Matrix's DSP supplier," said Eli Nakash, Surf's Sales Director, APAC. "This project demonstrates how Surf's multimedia processing expertise and dedicated customer care enable the rapid development of a variety of feature-rich, scalable solutions."

For more information, visit www.matrixtelesol.com.

Contact Information SURF Communication Solutions Eli Nakash, Director of Sales, APAC Tel: +972 (0)73 714 0700 e-mail: eli@surf-com.com
Matrix Telecom Pvt. Limited Ashvin Navadia, Sourcing Manager
Tel: +91 265 2630555 e-mail: Ashvin.Navadia@MatrixTeleSol.com

SOURCE: Surf Communication Solutions Ltd -0- 04/24/2008
CONTACT: Eli Nakash, Director of Sales, APAC, of SURF Communication Solutions, +972 (0)73 714 0700, eli@surf-com.com; or Ashvin Navadia, Sourcing Manager of Matrix Telecom Pvt. Limited, +91 265 2630555, Ashvin.Navadia@MatrixTeleSol.com
Web Site: http://www.surf-com.com
http://www.matrixtelesol.com

tag: Matrix Telecom, Surf DSP-24, VoIP gateway, IP PBX solution, VFX32 VoIP gateway, Voice Mail, Auto Attendant, Call Forward, Call Transfer, Call Park, Call Pick up, Do not Disturb, Hunting Call/Department Call, Time Based Call Routing, Number Based Call Routing, Least Cost Routing,

QubeConnect, VoIP IP Telephony solution based on open standards.

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.

Asterisk 1.6 Beta1 Ready For Testing

"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
tag: Asterisk, IP PBX, asterisk 1.6, asterisk beta, asterisk 1.4,VoIP Solution, chan_unistim, asterisk 1.6 beta1

Asterisk 1.4.17 released to fix SIP Security Issue.

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.

tag: Asterisk, SIP Security, cable VoIP, IP PBX,

Demand for IP PBX on the rise in business

According to an article on PC World, VoIP usage in the business' are on the rise. Both the United satates and in Europe.
Forrester Research recently surveyed some 516 landline voice decision-makers in North American and Europe to gauge their interest in IP telephony. Results show that more than half (54 percent) of those polled are increasing their budgets for IP PBX systems and services this year, with planned migrations most likely wrapping up in the next few years. The research firm attributes the growth to diminishing concerns over reliability and resiliency, as well as declining costs, and credits an ever-increasing mobile workforce as forcing more companies to consider IP telephony.
PC World article.

Tags: IP PBX, VoIP, landline, mobile workforce, IP telephony