Note:  This is a great post from the list.  Please if you attended take time to post your comments to this post.

For the benefit of those outside of the USA or those unable to make it to Astricon 2006;

I wanted to send out this email. For those of you who attended Astricon in Dallas last week what was the one thing that you saw that made the trip worthwhile? (if we post enough information or comments it will be of benefit for those that didn’t attend) For me personally it was the volume of neat add-on applications that the Asterisk community are developing; Over time I’m hoping that this leads to something like AppExchange from Salesforce.com were people can choose from over 300+ applications or addons for SF.

Posted by Dal at 09:37 AM | Direct Link Posted to Asterisk News

Posted by Dal at 08:27 AM | Direct Link Posted to WiFi / Wireless

 

“To achieve our vision of the ultimate guest experience, we needed a technology that made the guests more comfortable, made their experience more enjoyable, but also gave a practical benefit to the hotel in terms of operations,” said Brian Flaherty, general manager of Hotel 1000. “Cisco’s innovative, forward-thinking customer-oriented solutions were consistent with our effort to personalize our guest experience, thus enabling us to more effectively anticipate the needs of our guests and allowing us the opportunity to more effectively and efficiently deliver the service experience that they are expecting.”

The highly customized applications were jointly designed and deployed by Percipia, of Columbus, Ohio, a Cisco Technology Developer Partner, and Valcros, of San Diego, Calif., a Cisco Premier Certified Channel Partner. Cisco’s Smart Business Communications initiative helped align a flexible network technology plan with Hotel 1000’s top business priorities. As a result, the cost-effective converged platform supports the delivery of premium network services to guests, staff and management to help create a memorable experience that motivates guests to return.

“At Hotel 1000, Cisco Unified IP Phones are far more than a calling device,” said Chris Farrar, president of Percipia. “With Percipia’s hospitality applications, they are powerful interactive Unified Communication tools that deliver enhanced functionality while making a powerful statement about Hotel 1000’s commitment to a superior experience.”

“Hotel 1000’s owners and management had the foresight to design the information and communication infrastructure at the beginning of the building planning process,” says Mark Munger, CEO and president of Valcros. “That foresight enabled us to lay a foundation that maximizes hotel management and guest service delivery — from opening day into the future.”

Source: Cisco 

Posted by Dal at 08:07 AM | Direct Link Posted to PBX / iPBX

Note: Sean from i9 sent this over to me and looks interesting.  Below I have pulled a little info from there site.

It's my pleasure to introduce you to a new product for the open source project Asterisk. iSymphony is a cross-platform, real-time operator panel for Asterisk that comes in two flavors. A free downloadable version and iSymphony Conductor Edition.

iSymphony, an easy-to-use, Java-based client/server software for managing phone calls via i9's iPBX or the Open Source Asterisk platform. Due to Java's architecture, our software runs on many different computer operating systems and its user interface is clear and easy to master. download iSymphony displays:

* What extensions are busy, ringing or available

* Who is talking and to whom * Registration status and reachability

* Parked Calls Features:

* Hang-up a channel

* Transfer a call via drag & drop

* Originate calls via drag & drop

* Remotely monitor calls in real time from your PC (barge)

* Record calls * Mute/Unmute barged calls

* Per user authentication

At a quick glance, you will be able to see incoming calls and calls on-hold as well as the availability of all organizational personnel.

More Information Here:

www.i9technologies.com/isymphony

Posted by Dal at 10:50 AM | Direct Link Posted to Asterisk Software

As with the prior versions, it runs on the desktop of any Windows XP home or office computer. If you want a state-of-the-art phone system, look no further. Out of the box, it supports eight extensions and two lines with integrated voicemail and immediate email delivery of your incoming voicemail messages. To add additional extensions takes about 5 seconds.

And, yes, all your favorite Nerd Vittles applications are preinstalled and ready to go including weather forecasts for 1,000 airports, MailCall for Asterisk to read you your email messages, NewsClips for Asterisk to read you the news, the AsteriDex robodialer complete with a web interface to place your outbound calls and to serve up customized CallerIDs for your incoming calls, TeleYapper to broadcast reminders and messages to your clients or little league team, and our new GabCast (podcasting) Player for Asterisk.

Click Here for the Full Nerd 

Posted by Dal at 08:25 AM | Direct Link Posted to Asterisk Help

While we acknowledge this as pure fiction, the dream of unfettered communication is becoming more of a requirement in today's business world. Being mobile is more the norm than ever before. Use of mobile handsets to meet enterprise mobility requirements seems to be a partial answer, but only addresses one part of the overall requirements: off-campus connectivity. However, there's no seamless connectivity with the corporate information systems and often there's limited cellular coverage inside office or public buildings.

The advent of enterprise-wide Wi-Fi adoption has brought about the possibility of delivering a technology that bridges wireless carrier connectivity with on-campus WLAN connectivity—Fixed/mobile convergence (FMC). This term was coined by early mobility advocates of providing seamless bridging between Wi-Fi and cellular networks. These technologies, now in pilot testing, extend cellular connectivity to places where cellular coverage is weak or non-existent. Being able to begin a phone call in the wireless network and continue that call when traversing network boundaries will provide true mobility. Such evolving technologies bring the hope of mobility to users, but are not always adequate in addressing the full range of mobility requirements.

FMC alternatives FMC solutions, in principle, all propose supplying voice and data services based on seamless integration of cellular and wireless packet-switch technologies/services. All of the announced FMC solutions fall into one of two general categories, hosted either by fixed, wireless, or hybrid carriers (carrier-centric), or by enterprises (enterprise-centric). The major difference between these two approaches is where the application control point resides, and results in functionally distinct solutions. Carrier-centric solutions view Wi-Fi networks as a potential transport mechanism for traffic roaming between fixed and cellular networks, with application control remaining with the carrier/service-provider network

(Fig. 1). At its design core, an enterprise-centric solution presumes that the Wi-Fi and wired-network services are a single logical resource under the control of the enterprise, with application control being retained within the enterprise. Such a design allows for a more "native" architecture to be implemented where voice-over-IP (VoIP) can be supported as part of a solution complementing the mobility capabilities of the WAN component.

 

Posted by Dal at 08:08 AM | Direct Link Posted to PBX / iPBX

Posted by Dal at 08:02 AM | Direct Link Posted to Asterisk User Groups

"We look forward to expanding our offering and furthering our reach in the marketplace through this partnership with Polycom," said Mark Spencer, President and CEO at Digium. "The interoperability between Asterisk Business Edition and Polycom SIP phones ensures flexibility and broad support for Asterisk telephony features."

Polycom's award-winning SIP desktop and conference phones will be combined with Digium's Asterisk Business Edition, the professional-grade version of Asterisk, the industry's first open source PBX. Through the combined Digium and Polycom offering, SMB customers will have access to advanced telephony solutions which will be more affordable compared to proprietary systems. Additionally, the new offering will give customers the control, rapid feature development and deployment, and rich feature base that the Asterisk open source community and its partners provide. Per the agreement, Polycom will also be the preferred VoIP phone provider for Digium solutions.

"Open source PBX's are gaining significant momentum in the market place because they are easy to deploy and they offer partners and customers the ability to use it as a development platform for customized solutions," said Will Stofega, research manager, VoIP Services at IDC. "The combination of Polycom's broad family of high quality desktop and conference phones and Digium's low-cost, feature-rich platform will deliver a strong integrated VoIP telephony solution to the SMB market."

"The open source approach is an important part of the current IP PBX market and we want to support this community with the renowned voice quality and intuitive user interface of Polycom's phones," said Sunil Bhalla, senior vice president and general manager of voice communications at Polycom. "Through our co-development agreement, Polycom SIP phones will be easy to deploy and manage in a Digium Asterisk Business Edition environment and support the business telephony features to meet the needs of the SMB market."

As part of the agreement, Digium will modify the Asterisk graphic user interface (GUI), Asterisk Business Edition application and Asterisk OS to tightly integrate with Polycom's line of SIP-based desktop and conference phones. The co-development work will enable simplified provisioning and support for advanced telephony features like shared line appearance, XML microbrowser plugins and Polycom HD Voice. Polycom and Digium will also work together on joint marketing initiatives and joint selling through common channels. Polycom phones will also be the exclusive VoIP phone in Digium's Asterisk Appliance Developer Kit (AADK).

Digium Asterisk Business Edition

Asterisk Business Edition is the professional-grade version of Asterisk, the industry's first open source PBX. The Asterisk Business Edition provides tested reliability of critical functions and features, and is tailored for small and enterprise business applications. It supports up to 40 simultaneous calls with upgrades to 240 calls available. Asterisk Business Edition includes a supported rPath Linux distribution with an enhanced installer, an Asterisk technical manual and a quick-start guide, making Asterisk even easier to install, configure, and use.

Polycom's SIP-Based Phones

Polycom offers a complete line of SIP-based desktop phones, including the SoundPoint IP 650 with HD Voice, SoundPoint IP 601, SoundPoint IP 501, SoundPoint IP 430, SoundPoint IP 301, and SoundPoint IP Extension Modules. The line also includes the SoundStation IP 4000 which is a SIP-based conference phone within the award-winning, triangular-shaped SoundStation line, which have become an icon for voice conferencing quality in businesses around the globe.

Polycom's latest development, Polycom HD Voice, incorporates leading technologies that Polycom has developed through more than 15 years in voice communications, including wideband audio, enhanced signal processing, Acoustic Clarity Technology 2 (including next generation technologies for transparent full duplex, echo cancellation, dynamic noise reduction, automatic gain control and microphone management) and specialized system design to deliver unrivaled clarity and richness. This enables better clarity and improved intelligibility of information, which significantly improves comprehension and productivity and reduces listener fatigue. Polycom is working with Digium to make HD Voice available to Asterisk Business Edition customers.

Source: Polycom and Digium 

Posted by Dal at 11:09 AM | Direct Link Posted to Asterisk Hardware

 

 It is our responsibility to ensure exceptionally high standards of call quality and reliability to our call centers,” said Vance Dailey, vice president of information technology for Intermedi@ Marketing Solutions. “Ranch Networks was key in achieving those goals. I have been very impressed with Ranch Networks products, the professionalism and knowledge of their employees, and most of all their flexibility in working with us to provide the optimal VoIP solution.”

Specifically, Intermedi@ will be using the Ranch Networks RN 40 appliances throughout their call center platform. Intermedi@ Marketing Solutions is currently implementing a thousand seat unified VoIP PBX and predictive dialer based on the Asterisk and VICIDIAL open source projects across its five call centers. The RN 40 series of appliances feature the new VoIP Matrix Technology clustering technique developed to increase scalability, reliability and security across Asterisk server farms.

“Intermedi@ Marketing Solutions chose Ranch Networks because our technology supports large numbers of simultaneous calls without compromising reliability and quality of service,” said Dave Gombos, vice president of sales and marketing for Ranch Networks. “We provide reliable, redundant and uninterrupted VoIP service across the five call centers, while significantly reducing the load on their Asterisk servers.”

Posted by Dal at 10:20 AM | Direct Link Posted to Asterisk Hardware

Cisco Systems on Thursday is broadening its unified communications portfolio with plans to acquire mobile software vendor Orative in a $31 million deal.

Orative, a 4-year-old San Jose, Calif.-based startup, makes software that will enable customers to tie their cellular phones into their Cisco VoIP and unified communications deployments, said Alex Hadden-Boyd, director of mobile unified communications at Cisco, San Jose. Orative, which has 33 employees, is already a member of Cisco's Technology Developer Partner program.

Cisco customers will be able to use their cell phones to integrate with business communications applications, providing improved productivity, Hadden-Boyd said. For example, a caller could access presence information for employees in the corporate directory, enabling them to see at a glance whether users are available to take calls or messages. They will also be able to see and access messages from Cisco's Unity voicemail and unified messaging platform and interact with Cisco's Unified MeetingPlace voice and Web conferencing line.

In addition, businesses that deploy the Orative technology will now have a way to track and control cell phone usage, she said.

"Our unified communications channel partners will be able to add this to their portfolios. It will allow them to forge new partnerships with mobile operators and handset vendors," Hadden-Boyd said. Orative already has a few channel partners and had already begun recruiting Cisco solution providers, she said. The deal is expected to close in January, and products should be in channel partners' hands by the end of the first calendar quarter of 2007, she said.

Orative's product line includes the Orative Enterprise Server, which will sit inside the customer's firewall connected to Cisco's Unified CallManager IP-PBX, and Orative Client Software, which gets installed on users' cell phones over the air. The software runs on several mobile operating systems, including Blackbery, BREW, Java 2 Platform Micro Edition (J2ME) and Symbian OS. Orative is working to develop a version for devices based on Microsoft Windows, Hadden-Boyd said.

Carriers including Verizon Wireless and Cingular have already certified the technology for use on their networks, she said.

Source: InformationWeek 

Posted by Dal at 10:15 AM | Direct Link Posted to VoIP News

Posted by Dal at 07:27 AM | Direct Link Posted to Asterisk Software

Note: I have to be honest that I am on the fence about this whole Fonality/Trixbox movement.  Before I give me opinion I would like to say as soon as 2.0 goes to "stable" I will be testing it, focusing on its T1/E1 Interface support and how smoothly that goes.  Don't get me wrong I love the interface and usability work they have improved on.  I just am afraid if you make it too "one size fits all" and people don't support digium's work (and community) that it will slow down development on the flagship.  Before fonality go into the picture (Do they sell hardware or Consulting??) I did notice that trixbox was "not" as flexible as standard Asterisk.  If people really get use to the "plug and play" for there PBX they might not really get everything out of there phone system.  Anyways, here is a tidbit from my good ol' friend Tom and what here had to say:

"trixbox 2.0 beta will be available for download on Wednesday. This release will be Fonality's first big contribution to the trixbox/Asterisk community after the recent Fonality acquisition of trixbox. which certainly caused a stir within the Asterisk community. I spoke with Chris Lyman, CEO of Fonality, to find out more about this major new release of trixbox." 

First, I should point out that while previous version of trixbox have always been the easiest way to get Asterisk up and running in just minutes, trixbox 2.0 is much more than that.  First and foremost, trixbox 2.0 includes a new 'overall' web GUI to make the whole process "point and click". From this new web GUI you can simply select the modules you want (HUDLite, FreePBX, PHP, lame, etc.) and the web interface will automatically install them. Some of the packages are directly related to Asterisk such as HUDlite or FreePBX, while other options are ancillary, such as SugarCRM. The idea is you shouldn't have to know anything about the command line interface (CLI). In addition, many users wishing to install trixbox want to keep the server as unbloated as possible and not add any unnecessary modules/packages.

Click Here for the Full Story

Posted by Dal at 07:17 AM | Direct Link Posted to Asterisk News

Posted by Dal at 07:14 AM | Direct Link Posted to Asterisk Help

"We are pleased to have reached this significant industry milestone," said Vedat Eyuboglu, CTO of Airvana. "VoIP is a highly delay-sensitive application, and delivering VoIP over an IP-based wireless infrastructure requires sophisticated QoS techniques. Our demonstration of VoIP on Airvana's QoS-enabled commercial EV-DO Rev. A network infrastructure is a significant step towards commercial VoIP services over EV-DO Rev. A."

Specifically, Airvana made mobile-to-mobile VoIP calls using handsets based on QUALCOMM's MSM6800 baseband ASIC and software to establish QoS-based calls over delay-sensitive/low-latency flows in a commercial-grade Rev. A network. EV-DO Rev. A supports multiple QoS flows per handset each with its own unique QoS requirements. This capability allows VoIP and other multimedia traffic to receive higher priority over other best-effort traffic, such as web, email, and file transfer traffic.

"QUALCOMM is pleased to see Airvana's VoIP Lab succeed in this critical demonstration," said Roberto Padovani, CTO for QUALCOMM. "Rev. A promises to deliver a new breed of performance on CDMA2000 networks, and the VoIP Lab serves as a viable proving ground for operators preparing for Rev. A upgrades."

Source: Airvana 

Posted by Dal at 10:23 AM | Direct Link Posted to VoIP News

The Corraleta Technology SDK takes users from a web page to a VoIP call with a live customer service agent within 10 seconds. These calls are pure IP from the web browser to the customer contact centre PBX and are free, irrespective of the location of the caller or contact centre.

The Corraleta-developed application can be used to initiate calls to the contact centre via a button on a web page, or can trigger a callback from the contact centre to the user’s chosen telephone. Application examples include sales and support services for online merchants; emails embedded with click to call functionality; and pay-per-call online banner advertising

The Corraleta Technology SDK uses Java to deliver VoIP across multiple browsers and 3rd party white label applications, providing tight, platform-agnostic integration between traditional phone systems and core business applications, whether on the company internal network or on the Web. This contrasts with rival solutions that use ActiveX and work only with Windows platforms and the Internet Explorer browser.

According to Mexuar’s director Tim Panton: “The ability for web users to interact directly with customer service agents is proven to boost sales and overall service – especially in higher-value sales transactions. The Corraleta Technology SDK delivers click-to-talk 2.0 for businesses, enabling them to quickly deploy platform-neutral applications to streamline communications, enhance sales and service, and deliver communications cost savings.”

In addition to enabling IP calls, the Corraleta SDK can also delivers information about the user’s online experience to the merchant’s customer service agent when the call is initiated. The agent software can display this information directly or use it to trigger lookups in company databases to retrieve details such as customer records, purchase histories and so on.

At Astricon Dallas (www.astricon.net), Mexuar director Tim Panton will be giving presentations about the technology behind Corraleta and its ability to integrate Asterisk IP PBXs and web applications on Wednesday 25th Oct at 16:30; Thursday 26th Oct at 14:00 and Friday 27th Oct at 12:30.

Posted by Dal at 10:17 AM | Direct Link Posted to Asterisk Software

Posted by Dal at 10:11 AM | Direct Link Posted to Asterisk News

Posted by Dal at 10:08 AM | Direct Link Posted to WiMax

As additional features, chan_celliax can manage (on Linux) voicemodems that use the ALSA modem driver (but with voicemodems it can't interact with Skype), and has got CLI commands: console, dial and hangup similar to the ones in chan_oss, useful for testing without hardware.

chan_celliax runs on Asterisk 1.2 on Linux and Windows (with cygwin and a little modification to the compilation of Asterisk to avoid the calls to sigkill). At date it runs on Asterisk 1.4 only in Linux
(because of the different compilation procedure of Asterisk 1.4, that does not build on cygwin). Hopefully it will run on FreeBSD.

Together with chan_celliax is distributed the Celliax Developer's LiveCD, with a working installation of Asterisk, chan_celliax, and configuration utilities based on Knoppix. Celliax Developer's LiveCD comes complete with all the developer's tools needed to recompile from svn and remaster
the LiveCD itself.  The LiveCD contains also the cygwin installer and the tgz with the asterisk-celliax stuff to be untarred in a basic cygwin installation.

The lucky ones of you guys who are at Astricon Developer's Summit will find some cables and CDs in the Code Zone.

More info on the site www.celliax.org, with forums, downloads, svn, trac, etc.

Please, report bug and issues to www.celliax.org/trac , is pre-beta software ;-)

Happy hacking,

Giovanni Maruzzelli
 

Posted by Dal at 08:47 AM | Direct Link Posted to Asterisk Development

Note: I am tuned in currently and they are discussing some good stuff.  Tune In.

If you want to listen into the Developers Summit meeting at Astricon you can call 972-961-7666 or IAX2/conference@switch-2.nufone.net/4569.

Jeremy McNamara
 

Posted by Dal at 08:43 AM | Direct Link Posted to Asterisk Development

Posted by Dal at 01:31 PM | Direct Link Posted to Asterisk News

The certification is required by the Malaysian government to allow Sangoma to sell its solutions in the country and it ensures network security for end users. The national certification ensures that certified telephony products are compatible with the country's public network and telecommunications infrastructure.

“Malaysian and international markets are welcoming our telephony solutions,” says Sangoma Technologies president and CEO David Mandelstam. “These certifications add to the extensive portfolio of approvals Sangoma has already gained worldwide, which has eased the implementation and rollout of enabling technology-based products around the world.”

Certifications are valid and renewable over a one- year period and cover Sangoma's AFT-based A101, A102, A104 Quad and A104d.

Sangoma's AFT series cards have a uniform 2U form factor (120mm x 55 mm, 4.7" x 2.2"). The 3.3v/5v PCI card supports bus-mastering, ring-buffer DMA architecture and fully shared interrupts, making it ideal for demanding environments such as the Asterisk Open Source PBX/IVR project. “Malaysian-based Lanvik ICU Sdn Bhd (http://www.lanvik-icu.com/) helped us manage and carry out the certification process,” says Mandelstam. “We will now leverage our strategic relationships to strengthen Sangoma's market position in this region.”

Posted by Dal at 12:07 PM | Direct Link Posted to VoIP Hardware

"Vonage is pleased to offer its customers the convenience of 511 dialing to speed up their commute," stated Michael Tribolet, president of Vonage America Inc. "This is just another example of Vonage answering its customer's needs."

When subscribers dial 511, their call is routed from Vonage's network to their local 511 Advanced Traveler Information Service (ATIS). 511 calls are routed locally based upon the address the customer registered for their 911 service.

511 Availability

There are currently thirty 511 systems operating in 26 states. Most of these systems are statewide, however, some regional systems that cover separate metropolitan areas or regions in a number of states are also in place. Vonage has made 511 available in each of the states or regions where the service is available, as listed below:

    STATE              COVERAGE AREA

    Alaska             Statewide
    Arizona            Statewide
    California         Northern California/Sacramento region
    California         San Francisco Bay Area region
    Colorado          Statewide
    Florida             Southeast Florida region
    Florida             Central Florida/Orlando region
    Florida             Tampa Bay region
    Florida             Statewide system (covering all areas not included in regional systems)
    Idaho              Statewide
    Iowa                Statewide
    Kansas            Statewide
    Kentucky          Statewide (see OH for Northern KY)
    Maine              Statewide
    Minnesota         Statewide
    Montana           Statewide
    Nebraska          Statewide
    Nevada             Statewide
    New Hampshire   Statewide
    North Carolina    Statewide
    North Dakota      Statewide
    Ohio                Cincinnati/Northern Kentucky region
    Oregon             Statewide
    Rhode Island      Statewide
    South Dakota      Statewide
    Tennessee         Statewide
    Utah                Statewide
    Vermont            Statewide
    Virginia           Statewide
    Washington     Statewide
    Wyoming         Statewide

Source: Vonage Inc 

Posted by Dal at 09:47 AM | Direct Link Posted to Vonage

Posted by Dal at 09:42 AM | Direct Link Posted to SIP

Posted by Dal at 09:33 AM | Direct Link Posted to Asterisk Development

Posted by Dal at 09:33 AM | Direct Link Posted to Asterisk Development

Posted by Dal at 09:27 AM | Direct Link Posted to Asterisk Help

The second phase, already underway, which is expected to be completed in February, will continue to expand its own facilities-based network to areas including Colorado, Texas, New Jersey, Arizona, Washington, North Carolina and California. The Company plans, by the completion of the fourth phase of its expansion, expected to be finished by the end of 2007, to continue its expansion in 21 states to provide communications services to more than an available 200 million subscriber and enterprise lines, Wireless Broadband Providers, Internet Service Providers, Carriers and other Next-Generation Service Providers. When completed, VoIP expects to be one of the top five CLEC's based upon the size of its network.

"We are pleased to have created such a sizable and reliable network in a relatively short period of time," said Shawn Lewis, VoIP Inc.'s Chief Technology Officer. "In addition to offering cutting-edge VoIP technology, we are now building, what will be the one of the top five largest CLEC networks in the country, behind leading providers such as Level 3 Communications and XO Holdings. Our network has the ability to service a diverse customer base with an array of technologies and new applications, such as our Click4MeT brand Click-to-Call VoIP technology, as well as traditional voice and direct Enhanced 911 facilities. Within six months, we expect to more than triple the size of our network, providing our customers with even greater benefit and a more rewarding telecommunications experience."

Source: VoIP Inc. 

Posted by Dal at 09:24 AM | Direct Link Posted to VoIP News

Boston, Massachusetts has become the latest major U.S. city to give municipal Wi-Fi a try, unveiling plans for a pilot project covering 5000 homes in the Roxbury neighborhood.

This project was inspired by a city-sponsored Wireless Task Force study, which recommended that Boston establish municipal Wi-Fi networks in partnership with a non-profit corporation, with the aim of bringing the internet to under-served areas.

“We said we’d move quickly, and we have,” said the city’s Mayor, Thomas Merino. “The Boston Wireless Initiative is up and running.”

Two public hotspots have already been launched by the city, and construction of a mesh network will commence immediately, city officials noted, although no exact timeline was given.

Funding and resources will come in the form of donations from the community, as well as the contributions of ISPs and network vendors hoping for a shot at a citywide contract. So far, BelAir Networks, GigaBeam, and MetroNext have all supplied equipment for the network, while Verizon Communications, Cisco Galaxy Internet Services, and SkyPilot have contributed time and services.

Source: TeleClick 

Posted by Dal at 08:13 AM | Direct Link Posted to WiFi / Wireless

Posted by Dal at 08:07 AM | Direct Link Posted to WiMax

The “PacketSmart for Asterisk” solution has three distinct functions:

* Network assessment for VoIP

* VoIP troubleshooting

* Ongoing VoIP SLA monitoring

During the deployment planning phase, the micro-appliances can be used to simulate live VoIP traffic on the SMB network, to identify and fix problems before the Asterisk VoIP deployment. After Asterisk deployment, the Packet Island software agent that is left installed on the Asterisk PBX continues to collect detailed quality metrics that can be used by the SME and VAR for SLA monitoring and to isolate and troubleshoot transient VoIP quality issues. For multi-site deployments with multiple Asterisk PBXs, the software agents can be employed to do periodic network assessments to ensure that the inter-Asterisk connectivity continues to support good VoIP quality.

Netswitch, a leading technology management company based in San Francisco, has been using the ‘PacketSmart for Asterisk’ solution for Asterisk deployment planning at the US offices of a large multi-national company. “We looked at several VoIP management solutions in the market, but none of them matched the flexibility and ease-of-use we found in Packet Island’s solution. The micro-appliances are truly plug-and-play and can be easily moved around in the network to isolate congestion points and evaluate routes. Furthermore, with the software agents installed on the Asterisk PBX, we can now offer our customer a continuous monitoring service that is essential in staying on top of transient congestion issues that affect VoIP” said Stanley Li, CEO of Netswitch.

“Open source Asterisk is creating a huge disruption in the PBX marketplace. However, one of the big challenges in deploying Asterisk with VoIP today is in ensuring that the customer’s network is ready for VoIP before the Asterisk deployment, and to stay on top of transient VoIP quality issues that invariably crop up as the network evolves.

The problem with current VoIP assessment and monitoring solutions in the market is that they are more expensive than the entire Asterisk PBX. With ‘PacketSmart for Asterisk’ we are offering the industry’s first SaaS solution that has been specifically created for the VoIP lifecycle management needs of the Asterisk market. Instead of paying thousands of dollars up front, our customers pay us an annual service fee that works out to tens of dollars a month“ said Praveen Kumar, President of Packet Island Inc.

Posted by Dal at 07:49 AM | Direct Link Posted to Asterisk Software

A full list of improvements over version 1.2.1 can be found at:
http://queuemetrics.loway.it/news.jsp

QueueMetrics 1.3 allows data storage on both flat files and MySQL databases for bigger call centers. And of course comes with a 90-page user  manual that covers all aspects of it.

QueueMetrics is a commercial call center monitoring package, but is availabe free of charge for individuals, Asterisk hackers and small SOHOs.  You can request a trial key if you run a larger installation and would  like to test it in your own environment.

The latest version of QueueMetrics can be downloaded from:
http://queuemetrics.loway.it/download.jsp

Hope you like it,
l.
 

Posted by Dal at 07:14 AM | Direct Link Posted to Asterisk Software

Note that Asterisk 1.4 is not vulnerable to the chan_skinny exploit that resulted in updated releases of Asterisk 1.0 and Asterisk 1.2.

The team has also released Zaptel 1.4.0-beta2 and Asterisk-Addons 1.4.0-beta2; these releases contain only bug fixes and minor improvements.

As always, the release files are available on the Digium FTP servers at ftp://ftp.digium.com, in both tarball and patch file form. All of the release files have been signed with our GPG keys and the signature files are available in the same directories as the release files.

Thanks for using and supporting Asterisk!
 

Posted by Dal at 11:31 AM | Direct Link Posted to Asterisk Releases

This release also contains a number of bug fixes, and some improvements to the chan_sip channel driver (for SIP devices) to mitigate the impacts of a certain class of denial-of-service attacks that have recently been published.

All Asterisk 1.2 users are urged to update to this release if they use the chan_skinny channel driver, or to stop loading it if it is not needed ('noload=>chan_skinny.so' in modules.conf will cause this behavior).

The team has also released Zaptel 1.2.10, Asterisk-Addons 1.2.5 and libpri 1.2.5; these releases contain only bug fixes and minor improvements.

As always, the release files are available on the Digium FTP servers at ftp://ftp.digium.com, in both tarball and patch file form. All of the release files have been signed with our GPG keys and the signature files are available in the same directories as the release files.

Thanks for using and supporting Asterisk!
 

Posted by Dal at 11:14 AM | Direct Link Posted to Asterisk Releases

By connecting the router to computers equipped with RangeBooster N wireless interfaces, wireless performance can be maximised, allowing users to stay connected from virtually anywhere at home and in the office. The DSL-2740B can also be used with 802.11g and 802.11b wireless networks to enable significantly improved reception. The router also features the D-Link Click’n Connect Easy Install Wizard to help users set up their router with only a few clicks of the mouse. The DWA-547 RangeBooster N 650 PCI adapter is a Draft 802.11n (1.) client device that delivers increased wireless performance to the desktop PC. The adapter enables users to easily upgrade to the next generation of wireless technology and access their high-speed Internet connection while sharing photos, files, music, video, printers and storage.

This adapter will provide a faster wireless connection and is ideally placed for digital phone calls, an enhanced gaming experience, and more efficient downloading and video streaming. The DWA-142 RangeBooster N USB 2.0 adapter is a Draft 802.11n (1.) client device that delivers increased performance gains and wireless coverage that is up to 4 times farther than 802.11g technology. The adapter also has the added benefit of portability between notebooks and desktop PCs. A USB cable is supplied, which allows the position of the adapter to be adjusted for convenience and to gain maximum levels of reception.

With raw data throughputs of up to 270Mbps, these high performance wireless adapters are designed for greater wireless coverage in the home, enabling the signal to travel further and ideal for users that demand higher networking performance. Wireless performance is maximized by connecting the DWA-547 and the DWA-142 to a RangeBooster N 650 router or Access Point. Both adapters support WPA/WPA2 security and WEP data encryption to help prevent outside intrusion and protect personal information. “The launch of these new products further demonstrates D-Link’s commitment to Draft N,” explained Tahira Perveen, Country Sales Manager UK and Ireland, D- Link. “Offering a broad range of products makes it easier to enjoy the benefits of Draft N performance, in both the business and consumer environments.”

Source: Dlink 

Posted by Dal at 07:14 AM | Direct Link Posted to WiFi / Wireless

The combination of The Cloud (www.thecloud.net) and Skype (www.skype.net) is set to deliver the first mobile roaming Skype service in Europe. The Wi-Fi phones come from SMC and right now they can be employed in over 8,500 public hotspots.

The Cloud offers coverage at various locations that include airports, hotels and railway stations across the UK, Germany, the Nordics and The Netherlands. This service provider also operates large-scale networks across the City of London, Canary Wharf, Amsterdam and nine other UK cities.

After registering their phone for Skype via the Internet, customers can use the mobile phone to automatically connect to Skype when they are within range of one of the operator's 8,500 hotspots. The service will initially launch in the UK this month, with the Nordics and Germany to follow towards the end of the year. The service is due to commence in France, Spain, Italy and Benelux at the start of 2007. The first shipments of the phone are in English, with French, German, Portuguese, Italian and Spanish language version units coming in Q4, 2006.

Source: VON Mag 

Posted by Dal at 10:32 AM | Direct Link Posted to Skype

Hi All,

We are getting ready to release our Call Control Gateway application which allows for both remote phone control and PC to phone integration between LCS and an Asterisk PBX. The gateway is scheduled to be released in the beginning of Nov. Currently we are looking for Beta Testers that are interested in this solution. More information on the product, along with the Beta Application can be found on our website at:

http://www.m-networks.net/uccg/


William Mandra
M-Networks

Posted by Dal at 10:18 AM | Direct Link Posted to Asterisk Development

We need to determine a set of tests to run. It is my intention to simulate a typical enterprise/provider environment by having discreet components of the test operation - sip proxy (ser), sbc, gateway, soft-switch.

We will send various different types of common and uncommon traffic patterns to the test system, so we can measure and report on the performance of Asterisk on each component. I would also like to put Asterisk's SS7 implementation to some serious testing, but very honestly I have limited SS7 skills at this time. The Abacus also supports H.323 and SCCP, so we should also dedicate some time to these protocols as well.

Lets discuss,

Jeremy McNamara

Posted by Dal at 10:16 AM | Direct Link Posted to Asterisk Development

The INRIA office is outside of Paris, in Rocquencourt. There will be buses going to and
from downtown Paris, so we're staying in Paris to get proper inspiration :-)

My suggestion is that we start 10 AM on Monday, nov 20th, and continue to 3 PM (15:00) on Wednesday, November 22nd. On Tuesday, we're eating dinner together in Paris (sponsors welcome :-) )

This is going to be a very practical meeting with interoperability tests between various devices, SIP debugging and coding. We need to figure out a way to add proper handling of video attributes in Asterisk and maybe look at additional features I know that you are working on out there:

- Video on hold (streaming)
- Video prompts for IVRs
- T.140 text in addition to video
- Integration with 3G video
- Video conferencing
- Additional topics (codename-pinapple etc)

Let's investigate these areas together, trying to find solutions that we can work forward on integration with other Open Source products or just experience in connections to commercial products.

There will be a very limited amount of seats. You will have to cover your own costs for travelling and eating, but there will be no registration fee (thanks to INRIA)

**** Register by mailing info@edvina.net today.

We have no seats for people who wants to "listen in" or "meet people" 
- this is a working meeting only. I hope you understand this. You do not need to  be an
Asterisk guru or developer, but still have knowledge about SIP and Video so that you can contribute.

My travel agency will handle hotel reservations, so we can stay in the same place. They can also assist you with booking flights, if needed. Emnet, my admin, will take care of the details.

A big thank you to INRIA for hosting this meeting. And a big thank you to other french contributors that offered to help with meeting rooms.

Cheers,
/Olle & Philippe

Links:

INRIA's website :
http://www.inria.fr/index.en.html

Access INRIA Rocquencourt research unit :
http://www-rocq.inria.fr/en/inria-rocq/moyens_acces/index.htm

 

Posted by Dal at 09:02 AM | Direct Link Posted to Asterisk Development

Coordinated Systems, Inc., of East Hartford, CT, is pleased to announce the latest update to the Virtual Observer 3.0 Call Logger, a total call recording solution with support for most VoIP phone systems.

The Virtual Observer (VO) Logger can now be bolstered with virtual unlimited storage, enhanced playback and media encryption. The VO Logger is expandable to include a full featured quality monitoring suite. The VO Call Logger works in most VoIP, traditional TDM, and hybrid digital/analog/IP environments.

There are many call loggers on the marketplace today. Many of the more affordable solutions offer basic recording and limited playback features. Many are simple hardware recorders requiring managers to physically visit the unit to view recordings. Many offer extended features at a much higher price and will often force feature purchases that are neither necessary or in the future plans of the customer.

CSI’s approach is a bit different. As with their Virtual Observer 3.0 random sample recording edition, they pack all the important features into the core model and allow customers to pick and choose what other functionality they need. The VO Logger is priced for the mid-market, yet easily extends to the enterprise for multi-location logging to a centralized database. The VO Logger can also capture SMDR or CTI data for additional intelligence – as an optional add-on module. Certain configurations also allow for screen capture in a logger environment.

Similar to the aggressive pricing model for new customer solutions, the upgrade path for current customers of CSI is clear and inexpensive. Full upgrade releases are free for actively maintained customers and the only fee is the labor for the actual server conversion. This contributes, along with a devoted customer service team, to CSI’s 97.5% customer retention rate.

The VO Logger comes optionally equipped with two significant features: Auto-Archiving and Media Encryption. With Auto-Archiving, companies can achieve virtual unlimited storage using automated storage and purging rules combined with a DVD recorder, NAS or SAN device. Events can be stored to any network-enabled device.

The Media Encryption module automatically encrypts all recorded events – protecting your important customer information. This also is a mandate enforced by Visa’s new Payment Card Industry standard. The VO Logger with Media Encryption complies with this standard. Calls are de-crypted during playback for authenticated users. The VO Logger includes granular-level security controls to further ensure system data protection.

Virtual Observer supports a large variety of VoIP and TDM/Legacy phone systems: Cisco, Avaya, 3COM, Siemens, Nortel, Mitel, as well as any SIP-enabled VoIP systems.
Coordinated Systems, Inc. (CSI) has been in business since 1972.

CSI employs a "Start Small, Think Big" philosophy that allows call centers to use a phased approach and still receive a high impact return on investment when implementing voip call recording and quality monitoring technology.

Posted by Dal at 10:54 AM | Direct Link Posted to VoIP Software

Posted by Dal at 02:14 PM | Direct Link Posted to Asterisk Software

 

Posted by Dal at 09:47 AM | Direct Link Posted to Asterisk Help

Under this program, Verizon SIP Gateway customers opting for OCN-based metered plans can enjoy a reduced rate for calls from their end-users to select Verizon Business local customer telephone numbers (those utilizing legacy MCI local switches).

SIP is an Internet-based signaling protocol used for creating, modifying and terminating sessions between two or more users on the Internet, including Internet telephone calls, multimedia distribution and multimedia conferences.

“This is simply another way we’re making it easier for our wholesale customers to do business with us - by listening to them and continually evolving our offerings to help them help their customers,” said Tom Maguire, senior vice president, Verizon Partner Solutions.

Click Here to Continue Reading 

Posted by Dal at 10:35 AM | Direct Link Posted to SIP

 

The WiMAX transceivers minimize the need for additional components, reducing design time and speeding up time to market. Combining low power consumption, a noise figure of less than three decibels, as well as high adjacent channel rejection, the high-performance ICs deliver significant benefits to mobile, nomadic and fixed wireless access (FWA) equipment, says the company. "The past year has been extremely significant in the development of WiMAX, and recent trials have demonstrated the system's ability to provide wireless broadband over a vast area," said Ruud van den Brink, marketing manager, RF WiMAX product line, NXP. "As devices become smaller, so do the requirements for smaller components.

Fully-integrated circuits such as the UXF23480/60 are now essential to enable WiMAX to be incorporated into our customer's new products, as space on the PCB becomes a premium, and the need for additional components needs to be minimized." Some of the biggest challenges for WiMAX-enabled mobile devices are associated with handover from basestation to basestation. NXP has developed a high-quality solution which is proven to offer seamless handover.

In addition the UXF23480/60 is an interoperable solution which will work with a variety of basebands by utilizing standard analog I/Q and serial interfacing and co-exists with cellular, WLAN and Bluetooth standards. "NXP's family of WiMAX transceivers will make it easier for design engineers to bring new and inventive products to market. These robust transceivers will enable a wide variety of handheld devices to boost broadband communications. As demand for high-speed broadband access on mobile devices grows, NXP's family of WiMAX transceivers will help manufactures quickly integrate this capability into the next generation of consumer gadgets," said Ruud van den Brink.

Source: Electro Pages 

Posted by Dal at 10:31 AM | Direct Link Posted to WiMax

Enswitch is in production today on systems from tens to tens of thousands of live customers. More details (including links to the full list of features and a working demo of the web interface) are at:

http://integrics.com/products/enswitch/

New features in 2.3:

* Iotum integration. For more details, please see http://iotum.com/

* E911 provider integration.

* Simplified web interface for residential customers.

* SIP messaging support.

* Improved unlimited call IVR. This allows callers to transfer to any destination that costs you less than the revenue you earn from the inbound call. Users need no account. Allowed destinations are calculated in real time.

* Call back support for remote access.

* Rate plans can include minutes and SIP messages per month. Minutes and messages can optionally be rolled over for one month.

* Optional busy lamps component. Unlike the Asterisk "hint" mechanism, this works on a cluster with multiple Asterisk machines handling calls.

* Many smaller improvements based on customer feedback.
 

Posted by Dal at 01:49 PM | Direct Link Posted to VoIP Software

SightSpeed and AMD have collaborated to incorporate SightSpeed’s award-winning personal video communications services in the AMD LIVE! Entertainment Suite with a new video communications offering that is available free of charge from the AMD website (www.amdlive.com). The AMD LIVE! Communicator powered by SightSpeed offers innovative video and voice over Internet services that leverage the performance of the AMD LIVE! PC.

“Together SightSpeed and AMD are providing consumers a fun and simple way to stay in touch with friends and family from virtually anywhere,” said Aaron Feen, director of consumer marketing for AMD. “SightSpeed’s powerful personal video over Internet solution is an outstanding addition to the AMD LIVE! Entertainment Suite.” The collaboration between AMD and SightSpeed enables consumers to use their PCs and high-speed Internet connections to conduct video calls via webcam, communicating face-to-face at the touch of a few mouse clicks.

The AMD LIVE! Communicator also enables free PC to PC voice calls, the ability to place and receive calls to traditional landline and cellular phones (PSTN), pre-recorded video email messages as well as viewing downloaded television programming for personal use via SightSpeed TV. “We are very pleased that the SightSpeed technology has been chosen to be a key addition to the AMD LIVE! Entertainment Suite,” said SightSpeed CEO Peter Csathy.

“SightSpeed is considered by the experts in the media and analyst communities to be the best of breed available when it comes to personal video communications over the Internet. SightSpeed’s collaboration with industry leader AMD will expand the accessibility and opportunity to experience video communications to a far greater number of potential consumers throughout the world.” In addition to basic free services, consumers can also upgrade to premium services by visiting www.amdlive.com or by going direct to SightSpeed’s website (www.sightspeed.com). Premium services include robust capabilities such as:

o Unlimited Voice Mail Inbox

o Unlimited Video Mail Inbox

o Unlimited Multi-Party Video Conferencing with up to four simultaneous participants

o Multi-Party Text Messaging

o Personalized My SightSpeed Web Page

o Create and Save Video Mail Messages

o Detailed Call History Reporting

o Personal Video Mail Management Page

o Priority Technical Support The AMD LIVE!

Communicator is based on SightSpeed’s highly regarded latest product version, 5.0, which reflects the company’s central mission of providing the world’s best Internet video communications services. This application also includes SightSpeed’s recently released and enhanced video codec, which has raised the bar and set a new standard for personal video communications. After installation of the AMD LIVE! Communicator, users can receive enhanced video clarity across all network conditions, while continuing to experience full 30 frames per second video, no latency, and outstanding synch of video and voice unmatched by any other service.

Posted by Dal at 01:17 PM | Direct Link Posted to VoIP News

"Linksys is known in the industry as the global leader for IP, wireless and networking for the SOHO and Small Business market," said Bruce Chatterley, CEO of Speakeasy. "Linksys and Speakeasy customers perfectly intersect in their needs for a complete phone solution in both hardware and service. The attractive price point and rich feature set of the Linksys SPA 942 make it a highly appealing phone to add to Speakeasy's solutions portfolio. With the Linksys phone, we now have one of the strongest SOHO-focused VoIP bundles in the industry."

The Linksys SPA 942 is the best price/performance IP desktop phone that Speakeasy will sell and is ideal for a business using a Hosted IP telephony service. This Phone coupled with Speakeasy service will help minimize the cost of switching from an obsolete PBX infrastructure to a future-proof hosted telephony solution with significant business continuity advantages.

The Linksys SPA 942 allows individuals all the features of IP telephony and reduced long distance while working from a home or any virtual office. For example, consultants working at a client's location for extended periods or business teams working from home can simply plug the Linksys SPA 942 into any broadband connection anywhere in the world. Once connected, the employee shares the corporate office number, eliminates long distance charges, and enjoys popular Voice over IP features such as voicemail to email and the easy to use Microsoft Outlook Toolbar plug-in.

"Teaming with Speakeasy makes for a strong offering of VoIP hardware and service that was developed for the small business customer in mind," said Sherman Scholten, Director of Product Marketing. "Speakeasy shares our vision in making it easier to give small businesses a more complete VoIP solution that provides the features, benefits and capabilities that they need to run their company more efficiently. We look forward to working with Speakeasy and supporting their growth in the hosted VoIP market."

The Linksys IP Desktop Phone has a number of features that will provide small businesses with big business phone functionality including:

Speakeasy's retail price for the Linksys IP Desktop Phones SPA 942 will be under $220. The phone will be added to the Speakeasy Business VoIP service offering in early November 2006.

Posted by Dal at 10:59 AM | Direct Link Posted to VoIP Hardware

Infineon's Residential Gateway design comprises:

* Danube -- A highly integrated SoC (system-on-chip) based on a 32-bit 24KEc MIPS core, and integrating a VoIP co-processor, two integrated codecs, an 8/16-bit NOR/NAND controller, and an ADSL2/2+ (asymmetric digital subscriber line) transceiver supporting data rates up to 24Mbps. I/O includes USB 2.0, two Ethernet interfaces, SPI with DMA, serial UART with hardware flow control, JTAG/EJTAG, and 32 general purpose I/Os.

Click Here for More Information 

Posted by Dal at 10:40 AM | Direct Link Posted to VoIP Hardware

Meetings are held monthly on the second Saturday of each month, excluding July and December. The Agenda is posted online
http://www.voip-info.org/wiki/index.php?page=Twin+Cities+Asterisk+User+Group+Agenda

Meetings are held at Sound Choice Communications LLC... -= 7839 12th Ave So, Bloomington Minnesota USA 55425 =-
http://maps.google.com/maps?oi=map&q=7839%2012th%20Ave%20S%2055425

Come to a meeting to meet other asterisk users, see asterisk solutions,  win a door prize, eat food, or for the good company, to look for work, if your looking for employees, to go out for a drive, to get out of your house, whatever, JUST COME TO THE MEETING!

New visitors can help themselves to FREE FXO Interface cards (So you can connect your phone line, and have a timing source for meetme and IAX protocols). Some members have been known to swap hardware at the meetings.  Have extra VoIP gear, looking for VoIP gear? There's plenty of hardware to see. Have you been to a meeting recently?

Please come and share your own ideas and learn from others. As always, free food.

We are always looking for help with meeting topics. If you feel like taking the lead, please do and simply let me know if you need anything.

Meeting starts at 10:30am and parking is available in the rear of the building. Runs about 2 hours or less, and we'll order Pizza to the meeting for lunch. This month we will need to wrap up by 12:30pm or 12:45pm.

Look forward to seeing you there.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20User%20Group% 20TwinCities%20Minnesota%20USA

If you have a product or service you'd like to introduce to our members, send a private message to ejo1(at)soundchoicecomm.com and we'll see if we can't get you listed as next month's sponsor.

Sound Choice Communications is a reseller of Digium and Polycom products and we have inventory on hand. Give us a call and your items will be waiting for you on Saturday.  Thank you! +1.651-999-0888
 

Posted by Dal at 09:48 AM | Direct Link Posted to Asterisk Help

AstriCon 2006 Includes:

Posted by Dal at 11:30 AM | Direct Link Posted to Asterisk News

Note:  Fresh off the O'Reilly press.  Looks like an interesting book covering a range of topics from Voicecoder tricks to Quality of service issues.  I think I am going to mosie down to my local B&N to pick up my copy.  I wish OR made hard copy editions of these references.

Click Here for: VoIP Hacks

Posted by Dal at 04:59 PM | Direct Link Posted to VoIP News

 Note: After reading this release, this conference phone looks pretty sweet.  We will see if this helps the business case for Skype in the office.  If you use Skype in your business please post a comment or send me an email.  I would like to know how many of our readers use skype.
 
 
Polycom, Inc., leading provider of unified collaborative communications solutions, today announced that the Polycom VoiceStation 500 conference phone is Skype Certified. The integration of Skype and the VoiceStation 500 enables customers to easily host business quality conference calls in a small conference room or executive office using Skype. The VoiceStation 500 features Bluetooth wireless and an applications port that enables it to seamlessly work with Skype running on a PC.

"Skype estimates that nearly 30 percent of its more than 113 million Skype users are using the software for business purposes," said Stefan Oberg, vice president and general manager of desktop and hardware at Skype. "By collaborating with Polycom, we are able to offer Skype business users a Skype Certified device that features Polycom's renowned voice quality."

Featuring a compact, stylish design, the affordable VoiceStation 500 delivers Polycom's award-winning Acoustic Clarity Technology for full duplex, natural group conferencing that minimizes background echoes, word clipping and distortion to provide a high-quality voice conferencing experience. The VoiceStation 500 uses AC power and can connect to a standard POTS phone line, but it also features embedded Bluetooth wireless support and an applications port for connecting to a computer running Skype. These connectivity options can also be used to connect the VoiceStation 500 to other devices such as a mobile phone and other handheld devices. The VoiceStation 500 complements the previously announced Polycom Communicator, a portable, personal USB speakerphone device that delivers business-quality, high-fidelity wideband voice communications for Skype calls.

"The partnership between Polycom and Skype is a natural fit," said Will Stofega, research manager of VoIP services at IDC. "Skype is making inroads into business environments, which require business-quality solutions. Polycom is the leader in business quality conference phones. By working together, they are able to provide the growing legions of Skype business users with high quality products from a name they know and trust."

"The business world is becoming more global and more geographically dispersed, which requires communication solutions that can adapt to the way people need and want to work," said Sunil Bhalla, senior vice president and general manager of voice communications at Polycom. "Working with Skype to certify the VoiceStation 500 gives our customers an integrated solution that meets their mobility needs and provides an affordable group communications solution without sacrificing voice quality."

In addition to the Skype certified VoiceStation 500 and Polycom Communicator, Polycom offers a Computer Calling Kit option for its SoundStation2 and SoundStation2W conference phones, enabling those products to work with Skype. For more information about the VoiceStation 500, Polycom Communicator and Computer Calling Kit, please visit the Polycom website at www.polycom.com .

Posted by Dal at 10:29 AM | Direct Link Posted to Skype

 

With broad adoption of VoIP phone systems in the corporate sector, VAR's and reseller are quickly beginning to move from analog paging systems to networked paging devices that are easily connected to existing IP networks with a single Ethernet cable connection.

CyberData's VoIP Loudspeaker Amplifier provides an easy method for implementing SIP-based overhead paging systems in noisy environments such as warehouses and production areas. It is enclosed in a moisture-proof NEMA enclosure for outdoor applications. CyberData also offers a wireless option for remote locations.

The VoIP Zone Controller with 4-PortAudio-Out enables access to existing analog paging systems through a SIP-based VoIP phone system. The four audio outputs connect to a standard paging amplifier with audio inputs and supports up to 16 paging zones.

CyberData's VoIP Paging Server enables users to create software-defined zones for paging speakers in a VoIP phone network. The Web-based configuration tool provides a graphical interface to select individual speakers for paging zones in a SIP environment.

The Wall-mount Speaker Adapter allows for easy surface mounting of CyberData's VoIP Ceiling Speaker directly to a wall.

The VoIP Ceiling Speaker, Loudspeaker Amplifier, Paging Gateway and other VoIP devices will be at show in booth 700, V8 during the Internet Telephony Conference and Expo in the San Diego Convention Center, October 10-13, 2006.

Source: CyberData 

Posted by Dal at 10:23 AM | Direct Link Posted to VoIP Hardware

Skype on Tuesday issued an update that fixes a serious security flaw in its internet telephony software for Apple's Mac OS X.

A vulnerability exists in the way Skype for Mac handles web links, according to a Skype advisory. An attacker could construct a malformed Skype link which, when clicked on, can cause the application to crash or allow a system to be compromised.

The company said in its advisory: "A user of Skype for Mac who follows a specially crafted URL may experience a crash of the Skype software and possibly may execute arbitrary code without consent." The VoIP provider, part of online auction giant eBay, deems the issue "high" risk.

A miscreant could publish a malformed Skype link on a website, for example, and try to trick someone into following it, the company said.

The vulnerability exists in Skype for Mac releases prior to and including 1.5.*.79. It has been fixed in release 1.5.*.80 or later, which was available for download on the Skype website on Tuesday.

Source: Silicon 

Posted by Dal at 10:05 AM | Direct Link Posted to Skype

To be held Tuesday, October 10 at 1:15 pm, David discusses the real trends that SMBs and enterprises are now beginning to follow such as the rising adoption of PC-based/open source PBX solutions.

“There exists a new breed of Open PBXs/Switches such as Asterisk, FreeSwitch, and Yate that are pushing the boundaries of Open Source telephony,” says Mandelstam. “My presentation will discuss the main trends behind the migration from TDM to IP in the enterprise, and how this impacts voice solutions. I will also explore the transition that's occurring from hardware-based to software-based and what's behind it.”

Mandelstam's address is open to all registered attendees.

“Sangoma is widely recognized and respected as an industry leader and I am confident that our attendees will appreciate and value the opportunity to hear his perspective on Open Source PBXs,” adds TMC president and conference chairman, Rich Tehrani. “Widespread deployment of IP services has reinforced that this exciting technology is a viable, cost-effective communications solution. We look forward to educating senior-level executives about its long-term benefits at the show.”

“Being invited to speak at this prestigious conference is proof positive that our open source PCI solutions are now being recognized by the voice market worldwide,” adds Mandelstam. “I look forward to telling delegates about the real trends that SMBs and enterprises are now beginning to follow en-route to adopting a PC-based/open source PBX solution.”

Source: Sangoma 

Posted by Dal at 10:47 AM | Direct Link Posted to VoIP News

Posted by Dal at 03:49 PM | Direct Link Posted to Asterisk News

“By partnering with innovative companies like Critical Links we can offer the business community more options and features when building high-end telephony systems based on Asterisk,” said Jim Webster, director of software technologies at Digium. “We look forward to working together with Critical Links to expand the VoIP market and at the same time, communicate the important role open source will play in furthering the telecom market.”

“We are pleased to be partnering with Digium, the recognized world leader in developing Linux based VoIP technologies," said Joao Carreira, Critical Links CEO. "We have been incorporating their technology in our products since the start of 2005 and, as a result, the interest in edgeBOX has grown tremendously. We can now provide a complete voice and data networking solution in a single box which greatly simplifies VoIP deployment, is cost effective, and addresses the concerns that users have over security and call quality," he added.

Source: Critical Links 

Posted by Dal at 09:34 AM | Direct Link Posted to Asterisk News

Ok I admit it.  I have given into the web 2.0 trend that has taken hold on the web.  I love it.  Any trend that comes along and pushes usability improvements and user generated review and content, I am all for it.  Bring it on in large amounts.  I have added the "tag cloud" to Asterisk VoIP News today and I am impressed with the visual improvement achieved.  Please send in your comments if you like or hate this new addition.  Also if you don't see some other addition that we should add to improve your experience please let us know.  This site is as much the user’s site as it is our own.  Have a good week and enjoy the site.

Signed,

AVN Team 

Posted by Dal at 12:48 PM | Direct Link Posted to AVN Site Updates

Editor's Note:  Now this is very cool device.  I have been a wifi geek for some time now.  After reading this, I want to test a couple of these bad (hint hint).  I am interesting to see how well the mesh network technology works and if the transition to different access points is seamless.  At $449 a piece and the fact they have 400mw antennas makes it seem like a well valued product.

NomadISP today announced they have extended their product line with the latest in Recreation WiFi Technology. The new AH4 line of access points and NLOS (Non-Line-of-Sight) repeaters include 900mhz back-haul mesh network capability allowing them to be deployed in heavy tree and urban cluster environments, and advanced dynamic bandwidth management insuring consistent guest service regardless of users use.

Additionally, NomadISP extended their product line to include a low cost, high powered 400mw entry level WiFi system capable of delivering managed, monitored and bandwidth controlled WiFi to smaller venues at an affordable price.

"NomadISP’s continued innovation in technology allows our customers to deploy WiFi in an effective manner, insuring exceptional guest service. The latest AH4 product line solves the largest Public WiFi issues by offering a managed, bandwidth controlled solution in a cost effective package." said Kelly Hogan, NomadISP's Chief Executive Officer. "Owners of smaller locations can now have the same great service and guest support that NomadISP is known for, at a very affordable price."

The AH4 self installable entry level system starts at $449, and includes powerful features such as a standard 400mw WiFi radio capable of covering areas of several hundred feet, public access through controlled service, and a private WPA encrypted network for the venue owners. All standard Anywhere Hotspot features are available including the ability to charge for public access or offer it as an amenity, 7x24 monitoring, guest support, and automatic upgrades and maintenance. It can be expanded as needed to cover larger areas if desired.

NomadISP is the largest wireless service provider for the camping and recreational industry, servicing hundreds of locations across North America. NomadISP's custom manufactured hardware was designed specifically for the recreation industry providing RV parks, campgrounds and marinas with the most effective deployment of WiFi technology at the best price in the industry. NomadISP leads the industry in the development and deployment of commercial grade WiFi systems.

Posted by Dal at 10:58 AM | Direct Link Posted to WiFi / Wireless

Posted by Dal at 10:56 AM | Direct Link Posted to WiFi / Wireless

Setu ATA2LL can propagate the call release on the FXS in the form of CPC signal. The device senses this signal and frees the FXS port.  An FXS port can be programmed for any of the three CLIP protocols - DTMF, FSK ITU-T V.23 and FSK Bellcore 202.

The product provides a list of programmable numbers or part numbers with economical SIP account. This port can be used to dial emergency numbers and during unavailability of the Internet or power failures.

Call arriving from a SIP account can be routed to either one or both FXS ports. Matrix Setu ATA2LL supports PPPoE client and hence can be used with xDSL modems.  Dynamic allocation of SIP account is also possible using dial plan.

Posted by Dal at 10:45 AM | Direct Link Posted to VoIP Hardware