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April 28, 2009

Asterisk PBX 1.6.1.0 Now Available

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.1.0. Asterisk 1.6.1.0 is available for immediate download at
http://downloads.digium.com/pub/asterisk/

This is the first release in the 1.6.1 branch, which has additional features added since 1.6.0. Please see the CHANGES file for more information about the additional functionality

For those upgrading from previous versions of Asterisk, it is advisable to review the UPGRADE.txt file:
http://svn.digium.com/svn/asterisk/tags/1.6.1.0/UPGRADE.txt

Some highlights about changes in this release:
----------------------------------------------

* It is now possible to specify a pattern match as a hint. Once a phone subscribes to something that matches the pattern a hint will be created using the contents and variables evaluated.

* IAX2 encryption support has been improved to support periodic key rotation within a call for enhanced security.  The option "keyrotate" has been provided to disable this functionality to preserve backwards compatibility with older versions of IAX2 that do not support key rotation.

* res_odbc no longer has a limit of 1023 total possible unshared connections, as some people were running into this limit.  This limit has been increased to 4.2 billion.

* Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given adaptive capabilities.  What this means in practical terms is that if your realtime table lacks critical fields, Asterisk will now emit warnings to
  that effect.  Also, some of the realtime drivers have the ability (if configured) to automatically add those columns to the table with the correct type and length.

* Config file variables may now be appended to, by using the '+=' append operator.  This is most helpful when working with long SQL queries in func_odbc.conf, as the queries no longer need to be specified on a single line.

* Many many other changes that are too numerous to list here. See:

  http://svn.digium.com/svn/asterisk/tags/1.6.1.0/CHANGES


For a summary of the changes in this release, please see the release summary:

http://svn.digium.com/svn/asterisk/tags/1.6.1.0/asterisk-1.6.1.0-summary.txt


For a full list of changes in this release, please see the ChangeLog:

http://svn.digium.com/svn/asterisk/tags/1.6.1.0/ChangeLog


The following list of issues were resolved with the participation of the community, and this release would not have been possible without your help!

* Allow disconnect feature before a call is bridged
  - Closes issue #11583. Submitted by sobomax. Tested and additional coding by sobomax, dvossel, murf.

* Update app_fax to work with spandsp-0.0.6
  - Closes issue #13688. Reported by and patched by irroot.

* chan_h323 with H323Plus for TRUNK (SVN rev. 89183)
  - Closes issue #11261. Reported by vhatz. Patched by jthurman.

* Wrong usage of sscanf with use of uninitialized variable caused accidental
  parsing of RTP/SAVP
  - Closes issue #14000. Reported and patched by folke.

*  Realtime peers are never qualified after 'sip reload'
  - Closes issue #14196. Reported, tested, and patched by pdf.


Thank you for your continued support of Asterisk!

Posted by Dal at 06:14 PM | | TrackBacks (0) Posted to Asterisk Releases

April 24, 2009

New VoIP Network Enterprise Service Kit From Fluke Networks

 
 
Fluke Networks, provider of innovative Network SuperVision Solutions for the testing, monitoring and analysis of enterprise and telecommunications networks, announces the availability of the VoIP Enterprise Service Kit, designed to ensure successful deployment of VoIP phones over existing network infrastructure.
By using the three vital test tools included in this kit -- a cable qualification tester, an inline performance tester and a digital probe -- technicians can quickly eliminate the most common problems plaguing VoIP installations.

The VoIP Enterprise Service Kit fills a void created by existing test tools that only check cable integrity -- without looking at VoIP performance -- and tools that only look at the VoIP phone, without testing the cabling's ability to transmit voice traffic. The new kit reduces the risk of rework and call-backs by testing both infrastructure and phone performance while the technician is on site.

The VoIP Enterprise Service Kit is built around the CableIQ(TM) Qualification Tester. The CableIQ tester checks cable bandwidth to ensure it will support Voice Over IP requirements. This can prevent hours of downtime and troubleshooting when VoIP equipment is installed on cabling with insufficient bandwidth.

Also in the kit is the NetTool(TM) Series II inline tester with VoIP Testing Option. NetTool allows users to see into VoIP calls by placing NetTool between the IP phone and network. Users can now quickly diagnose phone boot-up and call control problems as well as measure key call quality metrics without the need of a costly, hard-to-use protocol analyzer.

The third fundamental tool in the VoIP Enterprise Service Kit is the IntelliTone(TM) probe. IntelliTone simplifies cable identification by responding to unique digital tones which are unaffected by sources of interference that hinder traditional tone-probe sets. Both the CableIQ Qualification Tester and NetTool Inline Tester generate digital tone that can be located by the IntelliTone probe, saving time during cable identification tasks.

The VoIP Enterprise Service Kit is one of several VoIP solutions offered by Fluke Networks. Earlier this week the company announced NetAlly VoIP Assessment and Troubleshooting Software, version 7.0. This new software package helps eliminate risk associated with deploying or expanding VoIP services by assessing the current state of the network and previewing the service before it is deployed on that network. Assessing the network, a requirement of many leading IP PBX manufacturers, makes VoIP deployments faster, more successful and less costly by reducing post-deployment troubleshooting.

Product availability

In addition to the CableIQ Qualification Tester, NetTool(TM) Series II inline tester and the IntelliTone probe, the VoIP Enterprise Service Kit includes six remote office IDs, used for identifying cable outlets at the far end of a link. The VoIP Enterprise Service Kit is available for immediate delivery through Fluke Networks sales partners worldwide.

Source: MarketWire

Posted by Dal at 10:40 AM | | TrackBacks (0) Posted to VoIP Hardware

April 22, 2009

Asterisk PBX 1.6.1.0-rc5 Now Available

The Asterisk Development Team is pleased to announce the fifth release candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc5 is available for immediate download at http://downloads.digium.com/pub/asterisk/

This release fixes a couple of issues with realtime music on hold that could cause Asterisk to crash, and an issue that caused hungup channels to stay up, leading to 100% CPU usage. Additionally, several minor issues and edge case scenarios have been resolved.

For a full list of changes in this release candidate, please see the ChangeLog:

http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc5/ChangeLog

Issues found in this release candidate can be reported at http://bugs.digium.com

Thank you for your continued support of Asterisk!

Posted by Dal at 11:31 AM | | TrackBacks (0) Posted to Asterisk Releases

April 21, 2009

Encrypting Voice Calls Between Offices and Mobile Cell Phones

Security vendor, Cellcrypt has announced an application which enables encrypted voice calls from smartphones to conventional office landline phones. The Cellcrypt PBX Gateway integrates with commercial office PBX systems and allows users to complete encrypted voice calls from existing landlines to mobile smartphones that are running Cellcrypt Mobile software.

The PBX Gateway also enables encrypted voice calls between landline locations that have PBX Gateway installed; for instance between offices in different locations.

Cellcrypt technology is currently undergoing certification to the FIPS 140-2 standard approved by the US National Institute of Standard (NIST).

“Organisations spend significant amounts of effort and budget securing their data but until now have not had a viable solution for voice data,” said Simon Bransfield-Garth, CEO at Cellcrypt. “While traditional secure voice solutions have provided poor call quality and fail across most international boundaries, Cellcrypt offers unparalleled voice quality, government-specification security and global coverage never before experienced, all using standard smartphone and PBX technology.”

Source: Cellular News

Posted by Dal at 03:01 PM | | TrackBacks (0) Posted to VoIP Security

Cut-rate prepaid plans shake up wireless industry

As wireless carriers start reporting first-quarter results this week, investors will be looking at the effects of some spectacular price cuts for prepaid cell phone service.

That's a change from recent years, when flashy new phones and data services hogged the spotlight. This year, the developments have been more appropriate for a recession: People who are least able to pay are getting cheaper service.

In traditional prepaid service, which generally has been marketed to people with iffy credit, customers buy minutes in advance, and often are charged a fee for each day they use the phone.

The big change this year has been the rise of prepaid plans with no limit on the minutes used.

In January, Sprint Nextel Corp. made a bold move to capture a larger share of the prepaid market, launching a service with unlimited calling, texting and Web access for $50 per month under its Boost Mobile brand.

The plan was partly a response to MetroPCS Communications Inc. and Leap Wireless International Inc., two upstarts building their own wireless networks. In the last few months, they've expanded into New York, Boston, Philadelphia and Chicago with unlimited plans that cost around $50 a month, depending on the options.

Click Here to Continue Reading

Posted by Dal at 02:57 PM | Posted to WiFi / Wireless

April 20, 2009

Cisco's new CCNA specializations - CCNA Security, CCNA Voice and CCNA Wireless

To become a specialized Cisco Certified Network Associate (CCNA), you must first be a "regular" CCNA, and then pass a single certification test in your specialist area.

There are three new CCNA specializations:

  • CCNA Security
  • CCNA Voice
  • CCNA Wireless

Each of these is an area of technology in which Cisco is pushing for a very strong presence.

CCNA Voice
The CCNA Voice certification ensures that you have the skill set to perform installation, operation, and administration of VoIP solutions. In preparing for the certification, you will gain a solid foundation in voice applications and their concepts, including Cisco Unified Communications architecture.

This specialization became available on June 24, 2008, and is also valid for three years. The prerequisite is a valid CCNA. The exam number and name that you will need is 640-460 IIUC (Implementing Cisco IOS Unified Communications). To learn more about this specialization, please see the official Cisco CCNA Voice page.

Click Here to Continue Reading

Posted by Dal at 09:28 AM | | TrackBacks (0) Posted to Cisco

AT&T unveils 2009 3G broadband expansion plans for Texas

AT&T Inc. outlined its 2009 wireless and broadband network expansion plans for Texas on Thursday. The Dallas-based company said it will expand its high-speed wireless 3G networks throughout the state, with a focus on rural areas, as well as its U-verse home broadband service.

AT&T did not say how much it will spend on this year's upgrades but said it spent more than $6 billion on infrastructure statewide from 2006 to 2008. It said its capital expenditures companywide for 2009 will total between $17 billion and $18 billion.

The highlights from the 2009 plans:

• The addition of 130 new cell sites in Texas, including in Austin, Dallas-Fort Worth, El Paso, San Antonio and Sherman-Denison, but with the majority of new sites in rural areas.

• The launch of 3G wireless data service in 11 Texas markets: Abilene, Amarillo, Beeville, Eagle Pass, Fredericksburg, Garner State Park, Giddings, Huntsville, Kerrville, Lufkin/Nacogdoches and San Angelo.

• Expansion of 3G in several Texas markets, including Austin, Dallas-Fort Worth and Houston.

• The launch of 850 MHz spectrum that will improve 3G wireless capacity and in-building coverage.

• Expansion of the AT&T U-verse network in Austin, Dallas-Fort Worth, Houston and San Antonio.

Source: Dallas News

Posted by Dal at 09:23 AM | | TrackBacks (0) Posted to VoIP News

Digium starts planning for the official Asterisk PBX Conference - AstriCon 2009

Editor's Note:  It's that time again.  Time to get ready for Astricon 2009, the largest official Asterisk Conference for this wonderful open-source PBX. 

Digium®, Inc., the Asterisk® Company, today released details about the sixth annual AstriCon Open Source Telephony Conference and Exhibition. The event brings together open source and telephony developers, systems integrators, entrepreneurs and Digium partners to discuss Asterisk, the most widely used open source telephony platform for creating custom communication solutions. Digium invites those who would like to speak at AstriCon to submit information for consideration by June 1, 2009, at www.astricon.net.

The event will take place from October 13-15, 2009, at the Renaissance Glendale Hotel and Spa near Phoenix, Arizona. Registration is now open and early bird rates are available until July 1, 2009.

AstriCon 2008 proved to be the open source telephony event of the year, attracting over 600 Asterisk enthusiasts, a record number of attendees, for three days of in-depth discussions. This year’s convergence of users, developers, resellers, entrepreneurs and other fans of open source technology will continue the celebration of one of the most influential open source projects with educational sessions devoted to the developing Asterisk ecosystem, trends in Asterisk use, the latest applications and a broad range of technical topics.

Click Here for More Information

Posted by Dal at 09:16 AM | | TrackBacks (0) Posted to Asterisk News

April 17, 2009

SEC goes after former VoIP Inc. executives for fraud

Three former VoIP Inc. executives are being targeted by the SEC for improper bookkeeping and lying to investors about the financial shape of the company.

A complaint filed Monday in Miami federal court says that between November 2004 and May 2005, ex-VoIP inc. CFO and VP of Finance Osvaldo Pitters and GM Terrell Kuykendall recorded $1.4 million in fake revenue from the alleged sale of computer hardware and fees for management services.
The inflated revenue raised the company's overall figures for 2004 by 43 percent.

It is also alleged that VoIP Inc. CEO Steven Ivester knew the company was struggling and the company's actual revenues were "substantially less" than its projections, said the complaint. He didn't question the company's financial statements, but resigned in October 2005 after unloading 4 million shares of the company's stock to make a tidy $4.4 million in profits. The SEC also says he didn't file the proper paperwork.

VoIP Inc.'s financial irregularities were discovered in March 2006, resulting in the resignation of Pitters and the subsequent firing of Kuykendall in April 2006.

The SEC is seeking for the three to lose the profits they made from their actions, pay a civil penalty and permanently bar them from acting as an officer or director of a publicly held company.

Source: Fierce VoIP

Posted by Dal at 03:39 PM | | TrackBacks (0) Posted to VoIP Politics

April 13, 2009

REVEALED! CrunchPad Web tablet is built by geeks, for geeks

Editors Note:  This is going to be the next killer piece of hardware.  Amazon better take notice, for its Kindle is going to be at the way side soon.  This tablet has a huge screen compared to it and it looks much less bulky (Not to say the Kindle is really bulky in the sense).   With a wifi connection and skype, this is the killer portable internet communications device.   I would like to see in Generation 2, a web cam built in that works with Skype so I can do video calls.  Kudos Mike and AlphaGeek, I will be getting one or you could send me one to review.  
 
News:
 
Spy pics indicate this US$300 touchscreen Web tablet created by TechCrunch founder and Web 2.0 entrepreneur Michael Arrington could even beat Apple to the ‘iTablet’ punch!
One of the most hotly-awaited tech products of 2009 has just broken cover. It’s not the oft-rumoured Apple tablet but it could be the next best thing, and potentially even more popular in its appeal to the geek elite.

It’s the CrunchPad, a touchscreen Web tablet created by alphageek and TechCrunch founder Michael Arrington.  The concept behind the CrunchPad is simplicity in itself: a slim and lightweight ‘Web tablet’ designed purely for the Internet. No keyboard, just a touch-sensitive 12 inch screen with the obligatory virtual keypad. Linux as the OS, of course, loaded onto a solid state drive for super-fast boot times and nimble performance.
 

Posted by Dal at 11:15 AM | | TrackBacks (0) Posted to Wireless Hardware

Panorama Antennas Announces a New Range of WiMAX Antennas

Panorama Antenna's range of WiMAX antennas are designed to provide efficient infill and improved terminal coverage for real world applications such as wireless internet access, machine to machine data transfer and smart metering. Panorama also offers a range of multiple antenna systems which can radically improve the performance of WiMAX networks operating MIMO technology.
"Panorama Antennas is a leader in providing high performance antennas for the global wireless communications market," stated Christopher Jesman, Managing Director. "The new WiMAX antenna range fits into our portfolio perfectly. With a variety of directional, omni-directional and MiMo antennas, Panorama is ideally positioned for the increased usage of these 4G technologies."
 
Panorama Antennas is a leader in providing high performance antennas for the global wireless communications market The new WiMAX antenna range fits into our portfolio perfectly. With a variety of directional, omni-directional and MiMo antennas, Panorama is ideally positioned for the increased usage of these 4G technologies.
 
For additional information on the new WiMAX range or to order samples visit http://www.panorama-antennas.com/wimax.

Posted by Dal at 11:10 AM | | TrackBacks (0) Posted to WiMax

April 07, 2009

Faxing for Asterisk brings enterprise grade functionality to Open Source PBX

Digium, the Asterisk Company, today announced Fax For Asterisk, a complete, cost-effective platform for the development of fax solutions. The offering provides Asterisk users and integrators a suite of user-friendly applications and a licensed version of the industry-leading fax modem software from Commetrex. To meet the demanding requirements of business users, Fax For Asterisk provides reliable faxing across the Internet and public switched telephone network (PSTN).
Asterisk is the most widely used open source telephony platform. The software is available free of charge and has been downloaded millions of times for use by individual developers and systems integrators creating custom telephony solutions for businesses. Asterisk is also available as the professional-grade and commercially supported Asterisk Business Edition.
 
"Asterisk users, developers and integrators now have a toolkit allowing them to integrate fax with their phone systems," said Bill Miller, vice president of product management at Digium. "With Fax For Asterisk, Digium offers a reliable and fully supported fax solution."
 
Fax For Asterisk interoperates with standards-compliant fax machines connected to Asterisk 1.4 and 1.6 on x86 Linux systems. It provides low-speed PSTN faxing via DAHDI-compatible telephony interface cards as well as VoIP faxing to T.38-compatible SIP end points and service providers. Fax For Asterisk operates at speeds up to 14.4kbps and supports V.17, V.27 and V.29 fax modems.
 
Fax For Asterisk is available free of charge from the Digium webstore at http://store.digium.com/ for one concurrent fax session. Multi-session licenses are available for a one-time fee of $38.50 per channel. Fax For Asterisk is available immediately. Fax capabilities for Digium's Switchvox IP PBX were announced in February of this year and are based on this solution. For more details, visit www.digium.com.

Posted by Dal at 12:31 PM | | TrackBacks (0) Posted to Asterisk News

Group urges FCC to open AT&T's 3G to Skype on iPhone

An advocacy group today asked the Federal Communications Commission to decide whether AT&T Inc. and Apple Inc. have broken federal rules by blocking iPhone owners from using the recently released Skype voice-over-IP (VoIP) service on AT&T's 3G wireless network.

"If you look at the consumer rights [spelled out] in the FCC's Internet Policy Statement, there is a chance that they might be violation by this practice," said Chris Riley, the policy counsel at Free Press, a Washington-based media reform group.

Riley today sent a letter to the FCC (download PDF) asking the agency to confirm that wireless networks must toe the line of the Internet Policy Statement, a set of rules adopted by the FCC in 2005 that guarantees consumers the right to access any online content on any device.

The group's letter cited the release Tuesday of Skype for Apple's iPhone as an example for the need to clarify the rules. Skype, which its maker said yesterday had been downloaded more than 1 million times since Tuesday, allows VoIP calls only via a wireless connection on the iPhone. However, iPhone users can't make VoIP calls on the carrier's own data network.

In the U.S., AT&T is the exclusive carrier for the iPhone.

AT&T claimed that it doesn't block VoIP traffic on its 3G network. "We do not prohibit VoIP," said company spokesman Mark Siegel. "But we expect our vendors not to facilitate the services of our competitors. We shouldn't have the obligation to promote our competitors."

Click Here to Continue Reading

Posted by Dal at 12:27 PM | | TrackBacks (0) Posted to Skype

April 03, 2009

Speakeasy Certifies Digium's Asterisk PBX for SIP Trunking

Speakeasy, a Best Buy company, has added Digium, the Asterisk company, to its growing portfolio of partners certified interoperable with Speakeasy’s expanded SIP Trunking integrated voice and data services.

Speakeasy further expands its SIP Trunking integration of voice and data services to reach an even larger SMB market with the certification of Digium’s Switchvox SMB and Switchvox SOHO IP PBXs and AsteriskNOW open-source telephony platform.

“We are excited to certify Digium for our expanded SIP trunking services,” said Bruce Chatterley, Speakeasy president and CEO. “By certifying Digium’s Switchvox and AsteriskNOW offerings, we are working together to provide solutions for small business customers to upgrade their telecom infrastructures regardless of their legacy voice and data systems.”

Digium is one of the first IP PBX manufacturers to be certified with Speakeasy. Speakeasy can deliver its SIP trunking service directly to Digium’s Asterisk telephony hardware products.

Source: Phone+ Mag

Posted by Dal at 02:18 PM | | TrackBacks (0) Posted to SIP

Digium offers paid support subscriptions for Asterisk PBX

Editor's Note:  This is big news and I am suspecting that they saw there efforts to try and force people into their closed-source Switchvox iPBX was actually losing revenue instead of embracing their open-source golden-child and providing support.  Don't get me wrong, Switchvox is a nice product and it does added value but going the licensing route and closed-source is what all the other companies (Cisco, Avaya, Shoretel, etc....) do and THAT is one of the most compelling reasons to go open source.  Just because customers are FAMILIAR with getting a license for their phone system DOESN'T MEAN THEY WANT ONE.   It might be a hard lesson for Digium to learn but they will get it, peolple don't not want to be locked. 
 
Digium announced at VoiceCon Orlando that it will support subscriptions for businesses using Asterisk, the open source IP PBX. The company, which formerly supported only commercial versions of Asterisk that it sold as packages, now offers four levels of service ranging in price from US$595 for a year to $7,995 for a year. A three-year commitment comes with a 10% discount.

Digium offers Asterisk for free download, but until now users had to rely on the open source community or other vendors for help.

The company says customers asked for the service, claiming that their CIOs were interested in the cost savings Asterisk could offer, but leery of lack of support.

The Level 1 service provides coverage for a single PBX server, two support cases per year, discounts on both training and attendance at the Asterisk conference, and response time of 48 hours. The top-tier Level 4 service includes coverage of 10 servers, unlimited support cases, an hour of consultation time and a response time of four hours on calls.

Customers can upgrade their Level 3 and 4 services to add coverage for more servers for US$495 and $395 each, respectively. Level 1 through 3 customers can buy additional support cases for $295 each. The services come without SLAs.

The service is available now.

Source: Computer World

Posted by Dal at 01:43 PM | | TrackBacks (0) Posted to Asterisk News

Grandstream Release the GXV3140 IP Multimedia Video Phone

 
The GXV3140's extraordinary video quality, advanced telephony features and rich multimedia applications distinguish this product in a unique class of its own. The GXV3140 splendidly blends state-of-the-art real time video conference capability with a number of popular web and social networking applications. It features an advanced 1.3M pixel tilt capable CMOS camera (with privacy shutter), a 4.3” 480x272 digital color LCD, dual Ethernet ports, SD & USB, a stereo headset jack, TV-out, stereo audio-out, and a full duplex speakerphone.
It supports a full HTML web browser, IM with Yahoo/MSN/Google, thousands of internet radio stations, popular online music networks such as Last.fm, Yahoo Flickr web photo album, personalized RSS feeds of news/weather/stock/currencies, calendar, alarm clock, and 9 languages (English, Spanish, Chinese, Japanese, Korean, French, German, Italian, Portuguese). The GXV3140 will support an even greater selection of music and photo networks as well as streaming from popular video sites in the very near future.

The GXV3140 raises the bar for personal multimedia communication by combining a number of cutting edge technologies. Such technologies include: a zero touch plug-and-play deployment, advanced audio/video streaming, hands-off personalization of news/weather/language/time based on automated detection of a users’ location, and 1-touch instant access to popular music/radio/photo/video sites and social networks.

The GXV3140 also incorporates a feature that will enable users to communicate with each other for FREE right out of the box through Grandstream’s industry leading peer-to-peer technology. The GXV3140 will boot up within minutes, ready for making video calls with ZERO configurations when the user plugs the device into a broadband router at home. With this smart peering plug-and-play technology, friends and family can connect with each other through high quality real-time video conferencing over any distance without a charge.

In addition to serving the consumer market, the GXV3140 presents an attractive solution for service providers (especially SIP trunk service providers). Internet Service Providers can offer video telephony and rich multimedia applications bundled with worldwide traditional PSTN termination and E911 service. The GXV3140 supports 3 independent SIP accounts, nearly all popular voice codecs, and leverages Grandstream’s field proven, broadly interoperable SIP technology.

“The GXV3140 is the most advanced and Web friendly IP multimedia phone on the market. By marrying social multimedia applications and IP video conferencing, it offers the consumer market an aesthetically appealing visual communications and entertainment experience,“ says David Li, CEO of Grandstream. “The GXV3140 provides a strong alternative to travel, allowing consumers or business users to stay closely connected with friends, family, colleagues and business partners located around the world for minimum cost. In today’s economic downturn, such a device is critical.

Consumers can maintain frequent visual contact with their loved ones while sparing their discretionary income. In addition, they can enjoy Web entertainment and social networking through 1-touch access on the GXV3140 without using a PC. Grandstream is very pleased to bring this innovative IP multimedia product and the tremendously rich social entertainment experience behind it to the consumer market.”

The GXV3140 will be generally available through Grandstream’s worldwide channel partners in early May for a manufacturer suggested retail price of $299.00 USD.

Posted by Dal at 01:33 PM | | TrackBacks (0) Posted to VoIP Hardware

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