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 recent