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    <title>Asterisk VoIP News</title>
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    <link rel="service.post" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1" title="Asterisk VoIP News" />
    <updated>2008-05-15T21:07:12Z</updated>
    
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<entry>
    <title>Thirdlane PBX 6.0 Simplifies Asterisk Management with Upgraded GUI</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/asterisk_software/thirdlane_pbx_60_simplifies_asterisk_management_with_upgraded_gui.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1653" title="Thirdlane PBX 6.0 Simplifies Asterisk Management with Upgraded GUI" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1653</id>
    
    <published>2008-05-15T21:02:40Z</published>
    <updated>2008-05-15T21:07:12Z</updated>
    
    <summary>Third Lane Technologies today announced a suite of powerful new capabilities for the Thirdlane PBX software appliance and Thirdlane PBX Manager GUI. The newest version of the company&apos;s open management platform for Asterisk, Thirdlane PBX 6.0 adds features for hosted...</summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="Asterisk Software" />
    
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        <![CDATA[<div align="justify">Third Lane Technologies today announced a suite of powerful new capabilities for the Thirdlane PBX software appliance and Thirdlane PBX Manager GUI. The newest version of the company's open management platform for Asterisk, Thirdlane PBX 6.0 adds features for hosted IP-PBX providers, VARs and end users. </div>]]>
        <![CDATA[<p align="justify">&quot;Thirdlane PBX is unique because it enhances Asterisk management without limiting its capabilities&quot; says Third Lane's Founder and CEO Alex Epshteyn. &quot;A proven field-tested GUI and new features like multi-site management, CRM integration and enhanced device auto-provisioning make Thirdlane PBX the most mature, feature-rich and flexible system on the market.&quot;  </p><p align="justify"><strong>A Proven Solution for Managing Asterisk PBX</strong>  </p><p align="justify">Thirdlane simplifies management of the popular open source Asterisk PBX, while giving system integrators access to the full Asterisk feature set for building and deploying telephony applications. VoIP service providers also benefit from PBX Manager's full multi-tenant capabilities for managing multiple virtual hosted Asterisk PBXs for small businesses.  </p><p align="justify">&quot;With the multi-tenant edition, customers can start with a hosted system,&quot; says James Sturtevant, CEO of Sigma Networks, a Sunnyvale, CA supplier of hosted and on-premises IP-PBX and networking systems. &quot;When they grow, all they have to do is install an on-premises PBX. </p><p align="justify">They can use the same phones. So it allows me to gain customers at an earlier stage and keep them through the entire lifecycle without additional training.&quot;  Since its debut in 2003 PBX Manager for Asterisk has proven itself as a robust and flexible platform for delivering business-class voice services, both on-premises and as hosted IP-PBX systems.  &quot;PBX Manager has been a core part of our business since the beginning,&quot; says Jon Warren, CEO of Business First Solutions a supplier of hosted IP-PBX services. </p><p align="justify">&quot;Almost 90 percent of our clients are call centers, many with multiple offices and remote employees. PBX Manager's Web interface lets clients manage all those locations centrally, and re-route calls on the fly and have the changes take effect immediately.&quot;  </p><p align="justify"><strong>A Competitive Solution for Service Providers</strong>  </p><p align="justify">Thirdlane PBX MTE 6.0 (multi-tenant edition) builds on this foundation to further enhance service providers' product portfolios, making it easier than ever to deploy and maintain the Asterisk PBX and hosted IP-PBX installations.  &Ograve;With enhanced conference room management, CRM displays, management of multiple servers using a single Web interface, multi-level IVR, advanced 'hunt' groups and more, PBX Manager 6.0 is just what we've been looking for,&quot; says George Stoutenburgh, CEO of Colorado-based AxisInternet, a national ISP and VoIP provider. &Ograve;We now have the system to replace the conventional PBXs and compete with larger national providers.  The new release also enhances Thirdlane PBX's value as a unified platform for deploying communications services and building seamless business processes that integrate voice. </p><p align="justify">Thirdlane delivers open source where it matters, explains Robert Messer, President of ABP Technology, a national value-added reseller network specializing in VoIP and IP communications. &quot;It provides a powerful, clean interface for users and administrators, while giving our Reseller Partners full access to the underlying Asterisk platform for configuring or creating customized IP Telephony solutions. They can plug-in their own modules without complicating support or upgrades.&quot;  </p><p align="justify"><strong>Thirdlane PBX 6.0 Highlights:  </strong></p><p align="justify">- Cluster management, enabling service providers and system administrators to manage multiple servers from a single management console with a single sign-on </p><p align="justify">- Enhanced auto-provisioning that supports more phones out-of-the-box and extends VAR's ability to customize the provisioning process. </p><p align="justify">- Enhanced conference configuration and management, delivering full-featured conferencing with real-time conference control through a rich Web interface </p><p align="justify">- CRM Integration with pre-configured connections to popular CRM solutions and search engines, plus the ability to custom-configure screen pops with any Web-based CRM </p><p align="justify"><strong>Availability:</strong> <strong><em>Thirdlane PBX 6.0 is available now</em></strong>.</p><p align="justify">Source: <em>ClickPress</em>&nbsp;</p><p align="justify">&nbsp;</p>]]>
    </content>
</entry>
<entry>
    <title>Asterisk PBX 1.4.19.2 Released</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/asterisk_releases/asterisk_pbx_14192_released.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1652" title="Asterisk PBX 1.4.19.2 Released" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1652</id>
    
    <published>2008-05-13T20:00:20Z</published>
    <updated>2008-05-13T20:04:00Z</updated>
    
    <summary><![CDATA[The Asterisk.org development team has released Asterisk version 1.4.19.2. This release includes some IAX2 channel driver updates.&nbsp; Asterisk 1.4.19.1 was released to address an IAX2 security vulnerability.&nbsp; Unfortunately, the changes to address the security issue had an unfortunate negative impact...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="Asterisk Releases" />
    
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        <![CDATA[<div align="justify">The <a target="_blank" href="http://www.asteriskvoipnews.com/cgi-bin/mt/Asterisk.org">Asterisk.org</a> development team has released Asterisk version 1.4.19.2. This release includes some IAX2 channel driver updates.&nbsp; Asterisk 1.4.19.1 was released to address an IAX2 security vulnerability.&nbsp; Unfortunately, the changes to address the security issue had an unfortunate negative impact on IAX2&nbsp; performance in Asterisk.&nbsp; These issues have been addressed and the related fixes are included in this release.&nbsp; The performance of IAX2 in Asterisk due to these changes should be far better than it was even before the changes were made for the security issue.<br /><br />Anyone that uses IAX2 should use this release instead of <em>1.4.19.1</em>.<br /><a target="_blank" href="http://downloads.digium.com/pub/telephony/asterisk/"><br />http://downloads.digium.com/pub/telephony/asterisk/</a><br /><em><br />Thank you for your continued support of Asterisk!</em></div>]]>
        
    </content>
</entry>
<entry>
    <title>Digium and Integrics Partner to Offer Open Source-Based Telephony Solution to Carriers</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/asterisk_news/digium_and_integrics_partner_to_offer_open_sourcebased_telephony_solution_t.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1651" title="Digium and Integrics Partner to Offer Open Source-Based Telephony Solution to Carriers" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1651</id>
    
    <published>2008-05-12T19:05:43Z</published>
    <updated>2008-05-12T19:07:55Z</updated>
    
    <summary><![CDATA[Digium, the Asterisk Company, and Integrics Ltd., a software engineering company providing telecommunications software to customers worldwide, today announced that they have partnered around sales and marketing of Integrics&rsquo; Enswitch, a carrier-grade switch based on the Asterisk telephony platform, to...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="Asterisk News" />
    
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        <![CDATA[<div align="justify">Digium<span />, the Asterisk<span /> Company, and Integrics Ltd., a        software engineering company providing telecommunications software to        customers worldwide, today announced that they have partnered around        sales and marketing of Integrics&rsquo; Enswitch, a        carrier-grade switch based on the Asterisk telephony platform, to        carriers worldwide. The agreement provides a route for closer technical,        sales and marketing collaboration between the companies and expands        Digium&rsquo;s involvement in the carrier market.</div>]]>
        <![CDATA[<p align="justify">       Digium is the creator and driving force behind Asterisk, the open source        telephony software deployed by more than 4 million servers worldwide to        manage voice over IP (<strong>VoIP</strong>) calls for businesses and individuals. More        resellers, telecom professionals and software developers choose Digium&rsquo;s        products than those of any other open source telephony company because        only Digium delivers the technical superiority, security and flexibility        associated with Asterisk. Asterisk powers Digium&rsquo;s        family of software and hardware appliances, including Switchvox&trade;,        Asterisk Business Edition&trade; and AsteriskNOW<span />.     </p><div align="justify">     </div><p align="justify">       Integrics sells Enswitch to telecommunications companies, Internet        telephony service providers (<strong>ITSPs</strong>), VoIP providers and others to allow        them to create offerings such as full-featured, telephone management and        billing solutions, calling card integration, toll free and number        translation services, voicemail, call queuing, automatic call        distributor, fax to email and multi-level interactive voice response.     </p><div align="justify">     </div><p align="justify">       Enswitch is based on Digium&rsquo;s Asterisk        Business Edition or open source Asterisk, based on each customer&rsquo;s        preference, as well as other open source software such as OpenSER, MySQL        and Linux. Enswitch runs on a single machine or on an Asterisk/OpenSER        cluster and supports high availability and failover, allowing for        failure of any single machine with only a few seconds of interruption to        service. The clustered architecture, which can be geographically        distributed, also allows additional machines to be added as the customer&rsquo;s        network grows.     </p><div align="justify">     </div><p align="justify">       &ldquo;Enswitch is the most integrated system        available today for hosted commercial telephony services,&rdquo;        said Alistair Cunningham, Managing Director at Integrics. &ldquo;There        are many other products available that provide PBX features, pre-paid        billing, post-paid billing, invoicing and so on, but only Enswitch        provides all of these in a single integrated product. Collaboration with        Digium will help us better capitalize on sales and marketing        opportunities and enhance our own product development over time.&rdquo;     </p><div align="justify">     </div><p align="justify">       &ldquo;The combination of Asterisk and Enswitch        provides a cost-effective alternative to traditional methods for        providing hosted PBX and other commercial telephony services,&rdquo;        said Steve Harvey, Vice President of Worldwide Sales at Digium. &ldquo;Asterisk&rsquo;s        open source architecture also lends itself to a high level of        customization to meet the needs of each customer. We believe that a        closer relationship with Integrics will help push Asterisk further into        the service provider market by giving those companies a very attractive        alternative to closed-source products.&rdquo;</p><p align="justify"><strong>Source:</strong> <em>Business Wire</em>&nbsp;</p>]]>
    </content>
</entry>
<entry>
    <title>Digium Doubling Annual Sales with Open Source VoIP</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/digium_doubling_annual_sales_with_open_source_voip.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1650" title="Digium Doubling Annual Sales with Open Source VoIP" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1650</id>
    
    <published>2008-05-09T19:52:41Z</published>
    <updated>2008-05-09T19:57:22Z</updated>
    
    <summary>We thought it would be good to give open source VoIP pioneers Digium equal time, given our recent post about Freeswitch. Digium&apos;s founder, Mark Spencer was the original author of the Asterisk PBX, one of the more mature open source...</summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
    
    <content type="html" xml:lang="en" xml:base="http://www.asteriskvoipnews.com/">
        <![CDATA[<p align="justify">We thought it would be good to give open source VoIP pioneers Digium equal time, given our recent post about Freeswitch. Digium's founder, Mark Spencer was the original author of the Asterisk PBX, one of the more mature open source VoIP platforms. The company's VP of Marketing, Bill Miller, told me that &quot;Last year was a big year for us; at the year-end we had finished our 24th straight profitable quarter. <br /></p>]]>
        <![CDATA[<p align="justify">We had grown to about 130 employees.&quot; While Digium doesn't disclose financial figures, its sales have reportedly been doubling over the past few years and it has at least $14 million in venture capital since launching.</p><p align="justify"> Digium has been traditionally focused on the SMB market with its line of telephony interface cards and small appliances and it is increasingly getting into services. Miller said, &quot;We position the products with 2-400 users as an IP PBX. That business is growing over 100 percent year-on-year.&quot; The company recently revamped its entire product line and has announced a distribution partnership with 3Com for the sub-30 user range.</p><p align="justify">Digium continues to invest heavily in the open source side of the Asterisk product as well. Miller told me that &quot;A sizable portion of our developers are 100 percent dedicated to the OSS side and don't do any commercial work.&quot; This is quite a difference from some other OSS-based companies like Red Hat that tend to cross-pollinate developers.</p><p align="justify">Digium has made some missteps in the past, which have stirred controversy among the community outside of the company. This was part of the genesis of several other forks of Asterisk. The genesis of much of the controversy has been the defense of the Digium brand, and Miller said, &quot;It's not Digium that's hurting OSS, it's the people who leverage the Digium brand and don't give back that hurt the OSS side of things. We value it with a vengeance.&quot; This is often a point of contention when companies form around OSS projects, and one that has played out in different ways over the years.</p><p align="justify">With regard to Freeswitch, Miller says that Digium doesn't &quot;see us competing with them any time in the near future.&quot; Freeswitch isn't currently capable of commanding the attention of the company in any big way. They are largely in different markets, with Freeswitch focused on service providers, and Asterisk focusing on the SMB market. The relationship is tenuous at best, but there is definitely room for many projects in the VoIP space.</p><p align="justify">Digium is in many ways a good model for building a company around an open source project. It is a profitable company that has grown very conservatively, picked experienced leadership, and has focused on products above all else. With sales doubling year-over-year, this is surely a company and a market segment to watch.</p><p align="justify"><a target="_blank" href="http://ostatic.com/161503-blog/digium-doubling-annual-sales-with-open-source-voip">Source</a></p><p align="justify">&nbsp;</p>]]>
    </content>
</entry>
<entry>
    <title>Google &amp; Clearwire uses WiMax to Pave Internet Autobahn</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/wimax/google_clearwire_uses_wimax_to_pave_internet_autobahn_1.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1649" title="Google &amp; Clearwire uses WiMax to Pave Internet Autobahn" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1649</id>
    
    <published>2008-05-08T22:32:42Z</published>
    <updated>2008-05-08T22:37:17Z</updated>
    
    <summary><![CDATA[&nbsp;&nbsp;Clearwire isn't the latest Google acquisition. The Internet search giant, though, has joined a group of blue-chip corporate investors in the new Sprint-Nextel bailout of Clearwire -- a move that will save WiMax and further Google's innovations in mobile search.&nbsp;Clearwire...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="WiMax" />
    
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        <![CDATA[<div align="justify"><img width="542" height="313" border="0" src="http://blog.searchenginewatch.com/blog/img/google%20clearwire%20wimax.jpg" />&nbsp;</div><div align="justify">&nbsp;</div><div align="justify">Clearwire isn't the latest Google acquisition. The Internet search giant, though, has joined a group of blue-chip corporate investors in the new Sprint-Nextel bailout of Clearwire -- a move that will save WiMax and further <a onclick="s_objectID=&quot;http://blog.searchenginewatch.com/blog/071212-154110_1&quot;;return this.s_oc?this.s_oc(e):true" href="http://blog.searchenginewatch.com/blog/071212-154110">Google's innovations in mobile search</a>.</div><div align="justify">&nbsp;</div><div align="justify">Clearwire and Sprint Nextel said today they plan to merge their wireless broadband units to create a $14.55 billion communications company. Sprint Nextel will own a majority equity stake (51 percent) in the new joint venture. <br /></div>]]>
        <![CDATA[<div align="justify">Clearwire, will receive a $3.2 billion cash infusion from Google Inc., Intel, Comcast., Bright House Networks and newly spun-off Time Warner Cable. The investment is based on a target price of $20 per Clearwire share and will give the companies a 22 percent stake in the new venture.  </div><div align="justify">&nbsp;</div><div align="justify">The new Clearwire JV will be headed by Ben Wolff, Clearwire's current CEO , who said in a statement that the merger's &quot;expanded relationships with Intel (INTC) and Google (GOOG) will expand our vision of an open network.&quot; He added that the partners will enables Clearwire &quot;to tap into some of the greatest innovators of our time.&quot;  </div><div align="justify">&nbsp;</div><div align="justify">Clearwire, a startup founded by cellular pioneer Craig McCaw, is shooting for a U.S. W9Max network of 120 million to 140 million people by the end of 2010.  So here's what we want to know: &quot;How fast will WiMax be?&quot;  Clearwire's first mobile WiMax network (being built in Portland) boasts speeds of 5 to 6 mbps on the downlink and 2 to 3 mbps on the uplink while going down the freeway.  Wow. That's not your father's Internet Highway.  That's the frackin' Internet Autobahn.</div><div align="justify">&nbsp;</div><div align="justify"><strong>Source:</strong> <a target="_blank" href="http://blog.searchenginewatch.com/">SE Watch Blog</a>&nbsp;</div>]]>
    </content>
</entry>
<entry>
    <title>Digium Expands Commitment to AstriCon</title>
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    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1648" title="Digium Expands Commitment to AstriCon" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1648</id>
    
    <published>2008-05-07T20:21:38Z</published>
    <updated>2008-05-07T20:25:08Z</updated>
    
    <summary><![CDATA[AstriCon, the industry&rsquo;s first conference and exhibit devoted to the most widely used open source telephony platform, Asterisk, is on track to be the largest and most successful in this, its fifth year. Digium, Inc., the creator of and corporate...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="Asterisk News" />
    
    <content type="html" xml:lang="en" xml:base="http://www.asteriskvoipnews.com/">
        <![CDATA[<div align="justify"><strong><em>AstriCon</em>,</strong>        the industry&rsquo;s first conference and exhibit        devoted to the most widely used open source telephony platform, Asterisk<span />,        is on track to be the largest and most successful in this, its fifth        year.<strong> </strong>Digium<span />,        Inc., the creator of and corporate sponsor of Asterisk, today        announced that the event will be held in Glendale, Ariz., near Phoenix,        from September 23-25, 2008, and invited submissions for presentations.        Digium&rsquo;s commitment to and investment in the        conference promises to make this year&rsquo;s        AstriCon the most educational of any Asterisk event, for expert users        and beginners alike.          </div>]]>
        <![CDATA[<p align="justify">       AstriCon is the pioneer and longest-running event devoted to all things        Asterisk, one of the most influential open source projects today.        Attendees will learn about trends in Asterisk use, the growing Asterisk        ecosystem, the newest applications and a wide range of technical topics        from Asterisk developers, users and entrepreneurs.     </p><div align="justify">     </div><p align="justify">       &ldquo;AstriCon is so vibrant and productive        because it brings together the people who live and breathe telephony        innovation,&rdquo; said Mark Spencer, creator of        Asterisk and Digium&rsquo;s chief technology        officer. &ldquo;This is the one time in 2008 when        the individuals who are most focused on development and use of Asterisk        will come together, making it the show that both enthusiasts and those        who are looking to get to know Asterisk must attend.&rdquo;     </p><div align="justify">     </div><p align="justify">       In the past, many Asterisk users and Digium partners needed to select        one Asterisk event to attend in the fall&mdash;AstriCon        or Digium|Asterisk World, which has been held in conjunction with        Pulvermedia&rsquo;s VON Conference &amp; Expo. However,        due to recent changes at Pulvermedia, Digium will not hold        Digium|Asterisk World this fall. As a result, the many ISVs, resellers        and integrators, and developers who selected one over the other in the        past, or needed to split attention and resources, will be able to focus        on AstriCon.     </p><div align="justify">     </div><p align="justify">       AstriCon 2008 will be held at the Renaissance Glendale Hotel and Spa in        Glendale, Ariz. Those wishing to submit speaking proposals may do so by        June 1, 2008. Registration is now open at <a target="_blank" href="http://www.astricon.net/">www.astricon.net</a>.     </p>]]>
    </content>
</entry>
<entry>
    <title>Clearwire and Sprint to (finally) form a wireless company</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/wimax/clearwire_and_sprint_to_finally_form_a_wireless_company.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1647" title="Clearwire and Sprint to (finally) form a wireless company" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1647</id>
    
    <published>2008-05-07T20:06:23Z</published>
    <updated>2008-05-07T20:21:09Z</updated>
    
    <summary><![CDATA[Editor's Note:&nbsp; Well I am glad they finally agreed on details for this.&nbsp; Clearwire is a good service (I use it) that really needs proper backing so it can expand into more markets and hopefully they can enhance the service...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="WiMax" />
    
    <content type="html" xml:lang="en" xml:base="http://www.asteriskvoipnews.com/">
        <![CDATA[<div align="justify"><strong>Editor's Note:</strong>&nbsp; <em>Well I am glad they finally agreed on details for this.&nbsp; Clearwire is a good service (I use it) that really needs proper backing so it can expand into more markets and hopefully they can enhance the service like having a map that shows me where there towers are so I can point my modem it the towers direction to reduce my latency.&nbsp; I hope someone from Clearwire reads this post.&nbsp; Contact me if needed.</em><br /></div><div align="justify">&nbsp;</div><div align="justify">Clearwire and Sprint Nextel are planning to merge their wireless broadband units to create a new $14.55 billion wireless communications company.</div><p align="justify">The new company, to be named Clearwire, will receive a $3.2 billion investment from Intel Corp., Google Inc., Comcast Corp., Time Warner Cable Inc. and Bright House Networks. The investment is based on a target price of $20 per Clearwire share and will give the companies a 22 percent stake in the new venture.</p>]]>
        <![CDATA[<p align="justify">Overland Park, Kan.-based Sprint Nextel Corp. will be majority owner with a 51 percent equity stake, while existing Clearwire shareholders will receive about 27 percent interest.</p><p align="justify">Clearwire, which will concentrate on rolling out a mobile network based on the emerging WiMAX standard, will also receive an investment from Trilogy Equity Partners, led by U.S. wireless industry veteran John Stanton.</p><p align="justify">WiMAX promises faster download speeds than the latest networks run by cell-phone operators, and it's even seen as a potential competitor to fixed-line broadband.</p><p align="justify">Rivals such as AT&amp;T Inc. and Verizon Wireless have eschewed WiMax, opting instead for upgrades to their current wireless broadband networks and a future technology called Long Term Evolution.</p><p align="justify">Clearwire already provides wireless Internet service in some parts of the country, using a WiMax-like technology. The company had a subscriber base of nearly 400,000 wireless broadband customers at the end of 2007.</p><p align="justify">The new company is looking for a U.S. network deployment between 120 million and 140 million people by the end of 2010.</p><p align="justify">Sprint and Clearwire, a startup founded by cellular pioneer Craig McCaw, had already announced their plans to build out networks using WiMAX technology, but had been looking for outside funding.</p><p align="justify">The new company will be led by Clearwire Chief Executive Benjamin Wolff, with Sprint Chief Technology Officer Barry West serving as president. West also leads Sprint's XOHM division.</p><p align="justify">The Kirkland, Wash.-based venture will house workers from Clearwire and Sprint's XOHM unit and will have research and development and other operations located in Herndon, Va. Its board will consist of 13 members at the start. Sprint will name seven of them, which will include at least one independent director. The investor group will name four members, including one independent. Eagle River, a private investment company controlled by wireless veteran Craig McCaw, will name one member, with the remaining independent member selected by Clearwire's nominating committee.</p><p align="justify">McCaw is expected to serve as non-executive chairman. Other anticipated board members include Sprint President and CEO Dan Hesse, Comcast Chairman and CEO Brian Roberts, Time Warner Cable President and CEO Glen Britt and Stanton.</p><p align="justify">The deal, which has been approved by the boards of all companies involved, is expected to close during the fourth quarter. The company will apply for a Nasdaq listing under the ticker &quot;CLWR.&quot; </p>]]>
    </content>
</entry>
<entry>
    <title>Introducing AIM Call Out for Asterisk</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/asterisk_software/introducing_aim_call_out_for_asterisk.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1646" title="Introducing AIM Call Out for Asterisk" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1646</id>
    
    <published>2008-05-06T18:44:59Z</published>
    <updated>2008-05-06T18:48:29Z</updated>
    
    <summary><![CDATA[Today we&rsquo;re taking a Margarita Break from our shiny new PBX in a Flash 1.2 server to play with AOL&rsquo;s new AIM&reg; Call Out. AOL actually introduced the service as an Open Voice API, but it walks and quacks like...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="Asterisk Software" />
    
    <content type="html" xml:lang="en" xml:base="http://www.asteriskvoipnews.com/">
        <![CDATA[<div align="justify">Today we&rsquo;re taking a Margarita Break from our shiny new PBX in a Flash 1.2 server to play with AOL&rsquo;s new AIM&reg; Call Out. AOL actually introduced the service as an Open Voice API, but it walks and quacks like a SIP termination gateway so that, of course, tempted us to try it. Since it is SIP-compatible, we thought it would be fun to see if we could get it working with Asterisk. It didn&rsquo;t take long...</div>]]>
        <![CDATA[<p align="justify">AOL Math: 1.7 + .3 = 4&nbsp;&nbsp; AOL has taken a page from Ma Bell in terms of creative mathematics. With each call, AOL first rounds UP the time of the call to the next minute and then rounds UP the total price to the next penny. Here&rsquo;s the way their Terms of Service describe it: </p><p align="justify">&ldquo;For point of clarity, the rounded up minutes are multiplied against the current rate effective at the end of the call (generally based on the location the call is placed) and then rounded up to whole cents (USD).&rdquo; So a 70-second call in the U.S. (which should cost under 2&cent; at 1.7&cent; per minute using Plain Old Math) actually is billed to you at 4&cent;. Charitably speaking, </p><p align="justify">it&rsquo;s creative to advertise the cost of a call in the U.S. as 1.7&cent; per minute with all the rounding that is taking place. For short calls, it can be more than double that rate once you factor in AOL&rsquo;s double rounding. In our example, the 70-second call is first rounded up to 2 minutes. And then the cost of the call is computed at 3.4&cent; for the already rounded up call. Then the 3.4&cent; computation is rounded up to 4&cent;. So you see 1.7 + .3 really does equal 4 in the bowels of AOL.</p><p align="justify"><strong>Source:</strong>&nbsp; <a target="_blank" href="http://nerdvittles.com/index.php?p=215">Click Here for the Full Nerd</a>&nbsp;</p>]]>
    </content>
</entry>
<entry>
    <title>Voice 2.0 Developers like Open Source</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/voip_news/voice_20_developers_like_open_source.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1645" title="Voice 2.0 Developers like Open Source" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1645</id>
    
    <published>2008-05-05T19:21:00Z</published>
    <updated>2008-05-05T19:24:31Z</updated>
    
    <summary>A survey of Voice 2.0 developers carried out by iLocus, a research firm focussed on emerging communications, reveals that 72% of them prefer to work with Open Source telephony platforms like Asterisk, OpenSER, and FreeSWITCH and offer services direct to...</summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="VoIP News" />
    
    <content type="html" xml:lang="en" xml:base="http://www.asteriskvoipnews.com/">
        <![CDATA[<div align="justify">A survey of Voice 2.0 developers carried out by iLocus, a research firm focussed on emerging communications, reveals that 72% of them prefer to work with Open Source telephony platforms like Asterisk, OpenSER, and FreeSWITCH and offer services direct to the consumer. The survey is part of a report &lsquo;Voice 2.0: 2008 Status Report&rsquo; published by iLocus today. </div>]]>
        <![CDATA[<div align="justify">Open Source platforms mentioned above are now considered carrier grade. For a standalone Voice 2.0 applications open source telephony platforms meet the developer criteria. Although working directly with telcos like BT (rather than going via vendors like Microsoft or Sylantro) is the second most favoured choice, it seems that Voice 2.0 developers overall prefer to take control of their development by utilizing open source platforms and then going direct to the end user. <br /> <br />Going direct to the end user may sound hip, but there are marketing costs involved. On the other hand there are clearly benefits in offering applications via platform vendor channels. To start with, the platform vendors have an established telco customer base, who in turn have paying customers which forms a natural first target population for a developer&rsquo;s Voice 2.0 application. With the carrier grade telecom platform the vendors are also able support a scalable deployment. <br /> <br />The survey also reveals that the Voice 2.0 developers are not so keen on consumer driven applications. While they might consider developing an application that can be utilized across both business and consumer segments, their preference is to develop applications that are used in the business world. This might be for monetization considerations. In the consumer segment it is hard to monetize the mashups. CRM is on the minds of three-quarters of the developers. Conferencing and mobile VoIP are the joint second most popular target </div><div align="justify">&nbsp;</div><div align="justify"><strong>Source:</strong> Webwire<br /> <br />Surprisingly SIP is the most popular API even with all the noise about web services APIs. Certainly some of the most popular Voice 2.0 applications are those developed by the ones with telecom background. How that will change over the next couple of years remains to be seen. But all the efforts around web services APIs then seem to make little sense if telcos/vendors are not able to attract web developers. </div>]]>
    </content>
</entry>
<entry>
    <title>Asterisk 1.4.20-rc1 Now Available</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/asterisk_releases/asterisk_1420rc1_now_available.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1644" title="Asterisk 1.4.20-rc1 Now Available" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1644</id>
    
    <published>2008-05-02T18:31:03Z</published>
    <updated>2008-05-02T18:34:38Z</updated>
    
    <summary><![CDATA[The Asterisk development team has released Asterisk version 1.4.20-rc1.This release is a release candidate for the upcoming official release of 1.4.20.&nbsp; It contains a large number of bug fixes over the previous release, 1.4.19.&nbsp; We would like to encourage the...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="Asterisk Releases" />
    
    <content type="html" xml:lang="en" xml:base="http://www.asteriskvoipnews.com/">
        <![CDATA[<div align="justify">The Asterisk development team has released Asterisk version <strong><em>1.4.20-rc1</em></strong>.<br /><br />This release is a release candidate for the upcoming official release of 1.4.20.&nbsp; It contains a large number of bug fixes over the previous release, 1.4.19.&nbsp; We would like to encourage the community to assist us in testing before we release 1.4.20.<br /><br />The release candidate is available on the download site.<br /><br /><a target="_blank" href="http://downloads.digium.com/pub/telephony/asterisk">http://downloads.digium.com/pub/telephony/asterisk</a><br /><br />Please provide release candidate testing feedback to the asterisk-dev mailing list, or the issue tracker, <a target="_blank" href="http://bugs.digium.com/">http://bugs.digium.com/</a><br /><br /><em>Thank you for your continued support of Asterisk!</em></div>]]>
        
    </content>
</entry>
<entry>
    <title>Zaptel 1.4.10.1 Released</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/asterisk_releases/zaptel_14101_released.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1643" title="Zaptel 1.4.10.1 Released" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1643</id>
    
    <published>2008-05-01T20:22:05Z</published>
    <updated>2008-05-01T20:23:29Z</updated>
    
    <summary><![CDATA[The Asterisk.org development team has announced the release of Zaptel version 1.4.10.1.&nbsp; This release is a bug fix release for a regression in which the Zaptel udev rules were not installed correctly, as well as a few minor fixes in...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="Asterisk Releases" />
    
    <content type="html" xml:lang="en" xml:base="http://www.asteriskvoipnews.com/">
        <![CDATA[<div align="justify">The Asterisk.org development team has announced the release of Zaptel version 1.4.10.1.&nbsp; This release is a bug fix release for a regression in which the Zaptel udev rules were not installed correctly, as well as a few minor fixes in the xpp drivers.<br /><br />This release is available as a tarball as well as a patch against the previous release.&nbsp; It is available for download from <a target="_blank" href="http://www.asteriskvoipnews.com/cgi-bin/mt/downloads.digium.com">downloads.digium.com</a>.<br /><br /><em>Thank you for your support!</em></div>]]>
        
    </content>
</entry>
<entry>
    <title>Yahoo to use Jajah for VoIP for 97 Million IM Users</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/voip_news/yahoo_to_use_jajah_for_97_million_im_voip.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1642" title="Yahoo to use Jajah for VoIP for 97 Million IM Users" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1642</id>
    
    <published>2008-04-29T18:24:52Z</published>
    <updated>2008-04-29T18:31:59Z</updated>
    
    <summary><![CDATA[Comments:&nbsp; I have followed Jajah for some time and I am very excited to hear this partnership with Yahoo!.&nbsp; I have talked to some of the staff and they all seem to be a really quality team and on the...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="VoIP News" />
    
    <content type="html" xml:lang="en" xml:base="http://www.asteriskvoipnews.com/">
        <![CDATA[<div align="justify"><strong>Comments:</strong>&nbsp; I<em> have followed Jajah for some time and I am very excited to hear this partnership with Yahoo!.&nbsp; I have talked to some of the staff and they all seem to be a really quality team and on the ball about VoIP technology.&nbsp; Grats<br /></em></div><div align="justify"><em>&nbsp;</em></div><div align="justify">JAJAH has been selected by Yahoo! as the outsource partner for its premium voice service. The &ldquo;Phone In&rdquo; and &ldquo;Phone Out&rdquo; service will enable consumers to make high-quality, low-cost PC-to-phone and phone-to-PC voice calls over the JAJAH network to more than 200 countries using Yahoo! Messenger, the leading instant messenger application in the United States with nearly 97 million users worldwide (comScore, February 2008).</div>]]>
        <![CDATA[<p align="justify">The partnership leverages JAJAH&rsquo;s open, next-generation communications platform to support the existing Yahoo! Messenger voice offering beginning in the third quarter of this year. Through this collaboration and managed service offerings, JAJAH will provide Yahoo! Messenger users with an instant Internet telephony network, merging the best of traditional and IP telephony.   </p><p align="justify">&ldquo;The seamless integration of web and voice interaction is clearly important to our strategic partner, Yahoo!,&rdquo; said Trevor Healy, JAJAH CEO. &ldquo;We are honored to be selected by Yahoo! to meet their voice needs. Through this relationship, we have the opportunity to extend our innovative global calling services with an industry leader that can leverage the power of our platform and network.&rdquo;  &ldquo;This partnership with JAJAH will help Yahoo! continue to provide a great communication experience to our millions of Yahoo! Messenger users,&rdquo; said Sabrina Ellis, vice president of Yahoo! Messenger. </p><p align="justify">&ldquo;Yahoo! Messenger is one of the first communication tools consumers see and use when they turn on their computers, so it is critical that our partner mirror our commitment to our users, and JAJAH&rsquo;s reliable VoIP network and proven customer- and carrier-friendly experience make it an ideal solution.&rdquo;  Since 2006, Yahoo! Messenger users have been able to use &ldquo;Phone In&rdquo; and &ldquo;Phone Out&rdquo; to make and receive voice calls on their PC to and from landline and mobile phones.  </p><p align="justify">With low rates and premium voice quality, users can talk for hours and save on their phone bill.  This deal means JAJAH will take over the provision of the telephony infrastructure, payment processing, and customer care for Yahoo!&rsquo;s premium voice users who make and receive voice calls through Yahoo! Messenger.</p><p align="justify"><strong>Source: </strong><a target="_blank" href="http://www.jajah.com">Jajah</a>&nbsp;</p>]]>
    </content>
</entry>
<entry>
    <title>Skype&apos;s New CEO &quot;Interview&quot; - He Still Doesn&apos;t Get It!</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/skype/skypes_new_ceo_interview_he_still_doesnt_get_it.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1641" title="Skype's New CEO &quot;Interview&quot; - He Still Doesn't Get It!" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1641</id>
    
    <published>2008-04-23T17:55:08Z</published>
    <updated>2008-04-23T18:03:43Z</updated>
    
    <summary><![CDATA[Editor's Comments:&nbsp; Good read, I love it when people take crusade against bad customer service.&nbsp;&nbsp; With us being part of this free market economy you really have two options, vote with your dollars (or euro if your into that) or...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="Skype" />
    
    <content type="html" xml:lang="en" xml:base="http://www.asteriskvoipnews.com/">
        <![CDATA[<p align="justify"><strong>Editor's Comments:</strong>&nbsp;<em> Good read, I love it when people take crusade against bad customer service.&nbsp;&nbsp; With us being part of this free market economy you really have two options, vote with your dollars (or euro if your into that) or take up the issues and not let the corporation get away with bad service and not hear what you have to say.&nbsp; Sometimes is takes a blog post, phone call or letter but make sure you get through to someone who can do something.&nbsp; </em></p><p align="justify">Skype has just published a so-called &quot;interview&quot; with their new CEO. First, rather than face the press and public, and possibly have to answer some hard questions from people who have (miserable) experience with Skype, he chose to hide behind a pseudo-interview with his own publicity manager. Second, even with every softball question possible being served up to him, it is obvious that he still doesn't get it.<br />&nbsp; <br /></p>]]>
        <![CDATA[<div align="justify">The biggest problem with Skype today is their disgraceful &quot;<em>Customer Support</em>&quot;. It is not simply lacking, it's worse than that, it's counter-productive. </div><div align="justify">&nbsp;</div><div align="justify">When someone is trying to set up or use Skype to talk to loved ones who are far away, to show Grandma the new baby, or to conduct important business, and it doesn't work (which happens all too often), first, they want to be able to contact Customer Support quickly and easily. </div><div align="justify">&nbsp;</div><div align="justify">Not through some obscure web page, which even makes it difficult to submit a help request, but by telephone, or at the very least by email. They want a response in a reasonable amount of time - and that is measured in hours, not days or weeks - and they want a response that is intelligent and has the potential to solve their problem, not something that is so patently ridiculous that they either laugh or cry over it.<br /> <br /> The second problem that absolutely must be solved at Skype is their habit of blocking users' accounts for no apparent reason (supposedly for the users own protection), and then taking days, weeks or even months to answer pleas for help and explanation.</div><div align="justify">&nbsp;</div><div align="justify"><a target="_blank" href="http://community.zdnet.co.uk/blog/0,1000000567,10007909o-2000498448b,00.htm">Click Here to Read the Rest of this Post</a>&nbsp;</div><div align="justify">&nbsp;</div>]]>
    </content>
</entry>
<entry>
    <title>Asterisk 1.2.28, 1.4.19.1, and 1.6.0-beta8 Released</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/asterisk_releases/asterisk_1228_14191_and_160beta8_released.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1640" title="Asterisk 1.2.28, 1.4.19.1, and 1.6.0-beta8 Released" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1640</id>
    
    <published>2008-04-23T17:34:51Z</published>
    <updated>2008-04-23T17:37:52Z</updated>
    
    <summary><![CDATA[The Asterisk development team has released versions 1.2.28, 1.4.19.1, and 1.6.0-beta8.All of these releases contain a security patch for the vulnerability described in the AST-2008-006 security advisory.&nbsp; 1.6.0-beta8 is also a regular update to the 1.6.0 series with a number...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="Asterisk Releases" />
    
    <content type="html" xml:lang="en" xml:base="http://www.asteriskvoipnews.com/">
        <![CDATA[<div align="justify">The Asterisk development team has released versions <em><strong>1.2.28</strong></em>, <em><strong>1.4.19.1</strong></em>, and<em><strong> 1.6.0-beta8</strong></em>.<br /><br />All of these releases contain a security patch for the vulnerability described in the AST-2008-006 security advisory.&nbsp; 1.6.0-beta8 is also a regular update to the 1.6.0 series with a number of bug fixes over the previous beta release.<br /><br />Early last year, we made some modifications to the IAX2 channel driver to combat potential usage of IAX2 in traffic amplification attacks.&nbsp; Unfortunately, our fix was not complete and we were not notified of this until the original reporter of the issue decided to release information on how to exploit it to the public.<br /><br />This issue affects all users of IAX2 that have allowed non-authenticated calls. For more information on the vulnerability, see the published security advisory.<br /><br />* <a target="_blank" href="http://downloads.digium.com/pub/security/AST-2008-006.pdf">http://downloads.digium.com/pub/security/AST-2008-006.pdf</a><br /><br /><strong>All releases are available for download from the following location:</strong><br /><br />* <a target="_blank" href="http://downloads.digium.com/pub/telephony/asterisk/">http://downloads.digium.com/pub/telephony/asterisk/</a><br /><br />Thank you for your continued support of Asterisk!<br /><br /></div>]]>
        
    </content>
</entry>
<entry>
    <title>Voxbone Awarded Licenses in Singapore and Greece</title>
    <link rel="alternate" type="text/html" href="http://www.asteriskvoipnews.com/voip_news/voxbone_awarded_licenses_in_singapore_and_greece.html" />
    <link rel="service.edit" type="application/atom+xml" href="http://www.asteriskvoipnews.com/cgi-bin/mt/mt-atom.cgi/weblog/blog_id=1/entry_id=1639" title="Voxbone Awarded Licenses in Singapore and Greece" />
    <id>tag:www.asteriskvoipnews.com,2008://1.1639</id>
    
    <published>2008-04-22T18:56:43Z</published>
    <updated>2008-04-22T19:03:04Z</updated>
    
    <summary><![CDATA[Voxbone announced that it has been awarded licenses and numbering resources to operate telecommunications services in Singapore and Greece.&nbsp; The new licenses, awarded by Singapore&rsquo;s IDA (Infocomm Development Authority) and Greece&rsquo;s EETT (National Telecommunications and Post Commission), bring the total...]]></summary>
    <author>
        <name>Dal</name>
        <uri>http://www.asteriskvoipnews.com/</uri>
    </author>
            <category term="VoIP News" />
    
    <content type="html" xml:lang="en" xml:base="http://www.asteriskvoipnews.com/">
        <![CDATA[Voxbone announced that it has been awarded licenses and numbering resources to operate telecommunications services in Singapore and Greece.&nbsp; The new licenses, awarded by Singapore&rsquo;s IDA (Infocomm Development Authority) and Greece&rsquo;s EETT (National Telecommunications and Post Commission), bring the total number of Voxbone-accessible countries to 43. <br />]]>
        <![CDATA[<div align="justify">Voxbone&rsquo;s DID (<em>direct-inward dial</em>) and toll-free numbers give its carrier, call center and major enterprise customers a way to establish a &ldquo;local&rdquo; presence in other countries, by being reachable through a local phone call.&nbsp; Seeing an advertised telephone number that costs little or nothing to dial, end customers and prospects will call an overseas businesses as soon as any business physically located nearby or in-country. Voxbone delivers these incoming calls to the number holder/subscriber through its global VoIP network.<br />&nbsp;<br />Customers can instantly lease and provision numbers through Voxbone&rsquo;s self-care Web portal, turning the acquisition process from months to minutes.<br /></div><div align="justify">&nbsp;</div><div align="justify">&ldquo;<em>Voxbone is our source for local access numbers whenever we extend the reach of our free- and low-cost international calling service,&rdquo; says Darren Yaphe, Director of Marketing at MOBIVOX, a Voxbone customer.&nbsp; &ldquo;We rely on those local access numbers, as they enable our customers in over 40 countries to reach our dial-by-name IVR every time.</em>&rdquo;<br />&nbsp;<br />Voxbone has partnered with local Singaporean and Greek telephone companies who are given their numbers by national regulators &ndash; a purchase as close to the physical source as possible and in compliance with all national standards and regulations. This assures number holders of service as reliable as that of a local incumbent carrier.</div><div align="justify">&nbsp;</div><div align="justify"><strong>Source:</strong>&nbsp; <a target="_blank" href="http://www.voxbone.com">Voxbone</a>&nbsp;</div><div align="justify">&nbsp;</div>]]>
    </content>
</entry>

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