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<channel>
	<title>VOIPSpeak</title>
	<atom:link href="http://www.voipspeak.net/feed/" rel="self" type="application/rss+xml" />
	<link>http://www.voipspeak.net</link>
	<description>The First and Last Word in VOIP</description>
	<pubDate>Sun, 10 Aug 2008 18:22:35 +0000</pubDate>
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	<language>en</language>
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		<title>trixbox CE - Creating an inbound dialplan / IVR menu</title>
		<link>http://www.voipspeak.net/2008/trixbox-ce-creating-an-inbound-dialplan-ivr-menu/</link>
		<comments>http://www.voipspeak.net/2008/trixbox-ce-creating-an-inbound-dialplan-ivr-menu/#comments</comments>
		<pubDate>Sun, 10 Aug 2008 18:19:59 +0000</pubDate>
		<dc:creator>admin</dc:creator>
		
		<category><![CDATA[Lead Article]]></category>

		<category><![CDATA[Asterisk]]></category>

		<category><![CDATA[design]]></category>

		<category><![CDATA[implementation]]></category>

		<category><![CDATA[IVR]]></category>

		<category><![CDATA[PBX config]]></category>

		<category><![CDATA[trixbox]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=91</guid>
		<description><![CDATA[Are you trying to get a grasp of how to create a good inbound dialplan when using trixbox CE? This article contains a video from our friends over at AsteriskTutorials.com that walks you through the design of a typical small - medium office call tree, how to flow chart the system, and then every aspect [...]]]></description>
			<content:encoded><![CDATA[<p><a href="http://www.voipspeak.net/wp-content/uploads/2008/08/trixbox.png" rel="thumbnail"><img class="alignright size-full wp-image-93" title="trixbox" src="http://www.voipspeak.net/wp-content/uploads/2008/08/trixbox.png" alt="" width="295" height="96" /></a>Are you trying to get a grasp of how to create a good inbound dialplan when using trixbox CE? This article contains a video from our friends over at <a href="http://asterisktutorials.com" target="_blank">AsteriskTutorials.com</a> that walks you through the design of a typical small - medium office call tree, how to flow chart the system, and then every aspect of programming it from within trixbox CE. This is a must-watch video for anyone getting started.</p>
<p><span id="more-91"></span></p>
<div align="center"><object width="579" height="437"><param name="allowfullscreen" value="true" /><param name="allowscriptaccess" value="always" /><param name="movie" value="http://www.vimeo.com/moogaloop.swf?clip_id=1480417&amp;server=www.vimeo.com&amp;show_title=1&amp;show_byline=1&amp;show_portrait=1&amp;color=00adef&amp;fullscreen=1" /><embed src="http://www.vimeo.com/moogaloop.swf?clip_id=1480417&amp;server=www.vimeo.com&amp;show_title=1&amp;show_byline=1&amp;show_portrait=1&amp;color=00adef&amp;fullscreen=1" type="application/x-shockwave-flash" allowfullscreen="true" allowscriptaccess="always" width="579" height="437"></embed></object><br /><a href="http://www.vimeo.com/1480417?pg=embed&amp;sec=1480417">Setting up trixbox CE inbound dialplans</a> from <a href="http://www.vimeo.com/kerryg?pg=embed&amp;sec=1480417">Kerry Garrison</a> on <a href="http://vimeo.com?pg=embed&amp;sec=1480417">Vimeo</a>.</div>
]]></content:encoded>
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		</item>
		<item>
		<title>FreeSWITCH 1.0.1 Release</title>
		<link>http://www.voipspeak.net/2008/freeswitch-101-release/</link>
		<comments>http://www.voipspeak.net/2008/freeswitch-101-release/#comments</comments>
		<pubDate>Fri, 25 Jul 2008 19:33:31 +0000</pubDate>
		<dc:creator>admin</dc:creator>
		
		<category><![CDATA[News]]></category>

		<category><![CDATA[Asterisk]]></category>

		<category><![CDATA[FreeSWITCH]]></category>

		<category><![CDATA[PBX]]></category>

		<category><![CDATA[telephony]]></category>

		<category><![CDATA[trixbox]]></category>

		<category><![CDATA[VOIP]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=89</guid>
		<description><![CDATA[We sat down with the folks at FreeSWITCH today to get a little behind-the-scenes info now that they have just released version 1.0.1 of their open source telephony platform. If you don't know much about FreeSWITCH, it was started as a project to be a telephony platform and designed from the ground up based on lessons learned during the development of Asterisk and hopefully avoid many of the problems that plague Asterisk today.]]></description>
			<content:encoded><![CDATA[<p><img class="alignnone size-medium wp-image-90 alignright" style="float: right;" title="freeswitch" src="http://www.voipspeak.net/wp-content/uploads/2008/07/freeswitch.png" alt="" width="295" height="68" />We sat down with the folks at FreeSWITCH today to get a little behind-the-scenes info now that they have just released version 1.0.1 of their open source telephony platform. If you don&#8217;t know much about FreeSWITCH, it was started as a project to be a telephony platform and designed from the ground up based on lessons learned during the development of Asterisk and hopefully avoid many of the problems that plague Asterisk today.</p>
<p>While FreeSWITCH wasn&#8217;t originally designed to be a simple PBX platform, the majority of contributions and requests revolve around PBX functionality, thus it has become a very powerful PBX platform as well as a soft switch. Some of the new features in 1.0.1 include:</p>
<ul>
<li>Polycom BLF support</li>
<li>Enhanced SNOM support for sidecards and BLF/SLA</li>
<li>mod_Flite to allow for text to speech</li>
<li>mod_pocketsphinx to allow creation of voice recognition within IVR&#8217;s</li>
<li>tons and tons of bug fixes and stability improvements</li>
</ul>
<p>What does FreeSWITCH lack today? Even the FreeSWITCH devs themselves will say the two major things missing today are documentation and examples. Every week the documentation and examples are being improved and this is making it easier and easier to get started with FreeSWITCH. Unlike Asterisk however, there is no web GUI for managing the PBX functionality at this time although rumour has it that there are a couple of development projects that are ramping up and hopefully we will see some basic systems within the next few months.</p>
<p>Is FreeSWITCH right for you? If you are comfortable with XML files and configuring your system by hand, FreeSWITCH sure seems to have all of the features you would need to build a production quality system. If the most experience you have with Asterisk is that you have installed trixbox and use the web gui to manage your system, then using FreeSWITCH today may be a little daunting. As more examples and improved documentation becomes available or when a good usable GUI comes out, then FreeSWITCH will start getting a lot more people trying it out.</p>
<p>From a developer point of view, FreeSWITCH is very contribution friendly as there is no dual licensing model or copyright waivers to sign. If you submit code and it passes their code review, it goes in and is available to everyone.</p>
<p>Be sure and keep an eye on FreeSWITCH, it may be the new player on the block but don&#8217;t underestimate its potential.</p>
<p>Website: <a href="http://freeswitch.org" target="_blank">http://freeswitch.org</a></p>
]]></content:encoded>
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		</item>
		<item>
		<title>BroadSoft and Fonality Team Up to Offer Managed IP PBX for Service Providers</title>
		<link>http://www.voipspeak.net/2008/broadsoft-and-fonality-team-up-to-offer-managed-ip-pbx-for-service-providers/</link>
		<comments>http://www.voipspeak.net/2008/broadsoft-and-fonality-team-up-to-offer-managed-ip-pbx-for-service-providers/#comments</comments>
		<pubDate>Wed, 16 Jul 2008 13:49:31 +0000</pubDate>
		<dc:creator>admin</dc:creator>
		
		<category><![CDATA[News]]></category>

		<category><![CDATA[Broadsoft]]></category>

		<category><![CDATA[Certification]]></category>

		<category><![CDATA[Compatibility]]></category>

		<category><![CDATA[Fonality]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=88</guid>
		<description><![CDATA[BroadSoft Certifies Fonality with BroadWorks, Fonality Welcomes                     BroadSoft into Its FACE Program
LOS ANGELES &#38; GAITHERSBURG, Md., Jul 16, 2008 (BUSINESS WIRE) &#8212; Fonality(R), the leading provider of open-source phone systems, and BroadSoft Inc., the leading [...]]]></description>
			<content:encoded><![CDATA[<h3>BroadSoft Certifies Fonality with BroadWorks, Fonality Welcomes                     BroadSoft into Its FACE Program</h3>
<div class="p">LOS ANGELES &amp; GAITHERSBURG, Md., Jul 16, 2008 (BUSINESS WIRE) &#8212; Fonality(R), the leading provider of open-source phone systems, and BroadSoft Inc., the leading provider of VoIP application software, announced today that the two companies have certified their products to work together, enabling service providers to market and deploy Fonality to their small and medium-sized business (SMB) customers with confidence. BroadSoft(R), which provides VoIP applications and SIP trunking to seven of the top 10 and 13 of the 25 largest carriers worldwide, has completed certification of Fonality trixbox(R) Pro and PBXtra(R) with BroadWorks(R), BroadSoft&#8217;s industry-leading VoIP application platform for fixed-line and wireless service providers. BroadSoft has also joined the Fonality Authorized Certified Ecosystem (FACE).
</div>
<div class="p">BroadWorks offers a range of carrier-grade applications that includes hosted PBX, unified communications, mobile PBX, business trunking, and residential broadband. Fonality products include a family of open-source-based, hybrid-hosted IP PBX offerings tailored for SMBs. By certifying the products together, the companies offer service providers a complete line of hosted and premise-based unified communications offerings for customers of all sizes.</p>
<p>&#8220;By collaborating with Fonality, we&#8217;re giving service providers a way to go to market quickly with fully integrated, market-tested offerings for smaller companies,&#8221; said Leslie Ferry, Vice President of Marketing for BroadSoft. &#8220;Service providers can now take new products to their customers with the confidence that comes from knowing that both companies&#8217; solutions will work together seamlessly.&#8221;</p></div>
<div class="p">
&#8220;BroadSoft is the dominant VoIP platform deployed by service providers and MSOs,&#8221; said Chris Vuillaume, Vice President of Business Development and Channels at Fonality. &#8220;Their certification is an important stamp of approval for Fonality products and expands our market opportunity to include Tier 1 and Tier 2 providers.&#8221;</div>
<div class="p">
Fonality business phone systems are designed for modern workplaces, accommodating companies that have a mix of office, mobile and home-based workers. Fonality solutions support both VoIP calling and traditional phone lines, allowing a smooth transition for businesses upgrading their calling services. Its patented, hybrid-hosted architecture allows employee identity to be maintained as they travel between work, home and hotels. Fonality products, when paired with the award-winning HUD(R) presence software, provide a unified view of Instant Messaging, e-mail and calling for all fixed and mobile workers.</div>
<div class="p">About the BroadWorks Platform</div>
<div class="p">
BroadSoft&#8217;s IMS-compliant BroadWorks(R) platform provides a comprehensive range of VoIP applications, including Hosted Unified Communications, Mobile PBX, Business Trunking and residential broadband services fully integrated into a single VoIP application platform. BroadWorks is capable of providing these applications with the reliability, redundancy, scalability and regulatory capabilities required to deliver carrier-class service.</div>
<div class="p">About BroadSoft</div>
<div class="p">
BroadSoft(R) provides VoIP application software that enables the delivery of hosted telephony and multimedia services. Its award-winning flagship BroadWorks technology empowers wireless, wireline and cable carriers to deliver next-generation voice and multimedia applications and advanced features that enable them to increase revenue, enhance competitive differentiation and elevate customer satisfaction. BroadSoft&#8217;s family of carrier-class software products delivers the scale, open architecture and reliability that the world&#8217;s leading telecommunications companies demand to serve mission-critical enterprise and residential broadband customers. BroadSoft provides VoIP applications to seven of the Top 10 and 13 of the Top 25 largest carriers worldwide, as measured by recent annual revenue, including Korea Telecom, KPN, SingTel, Sprint, Telefonica de Espana, Telstra, T-systems and Verizon. For additional information, go to  <a class="lk001" href="http://www.broadsoft.com/" target="_blank">www.BroadSoft.com</a>.</div>
<div class="p">
<strong>About Fonality </strong></div>
<div class="p"><strong> </strong>Fonality,  <a class="lk001" href="http://www.fonality.com/" target="_blank">www.fonality.com</a>, is a leader in business phone systems and contact center solutions for small and medium-sized businesses. Used by over 5,000 companies and 100,000 end users in 100 countries, Fonality&#8217;s award winning IP-PBX VoIP phone systems have connected more than 225,000,000 mission critical phone calls. The PBXtra and trixbox Pro product lines are based on Fonality&#8217;s patent-pending Anywhere Management(TM) Hybrid-Hosted(TM) architecture, plus an improved version of the popular open source Asterisk code base which has been modified to add reliability, stability and enterprise-class features. PBXtra and trixbox Pro deliver the advanced capabilities of an enterprise-class phone system for 40 to 80 percent less than traditional offerings. Fonality&#8217;s fully free and open source telephony platform, trixbox CE ( <a class="lk001" href="http://www.trixbox.org/" target="_blank">www.trixbox.org</a>), is home to one of the world&#8217;s largest and fastest growing communities of open source telephony users, with over 200,000 live deployments and 125,000 new downloads each month. Fonality headquarters are in Los Angeles and company investors include Azure Capital Partners and Intel Capital.</div>
<div class="p">BroadSoft and BroadWorks are registered trademarks of BroadSoft, Inc. Fonality, PBXtra, trixbox, and HUD are registered trademarks and Hybrid-Hosted and Anywhere Management are trademarks of Fonality. All other names are trademarks of their respective companies.</div>
<div class="p">SOURCE: Fonality</div>
<pre>BroadSoft
Francis Hopkins, +1-240-364-5375
fhopkins@broadsoft.com
or
Spark PR for Fonality
Phil Novack, 415-321-1870
phil@sparkpr.com
</pre>
]]></content:encoded>
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		</item>
		<item>
		<title>Yeastar announces Windows-Based PBX System</title>
		<link>http://www.voipspeak.net/2008/yeastar-announces-windows-based-pbx-system/</link>
		<comments>http://www.voipspeak.net/2008/yeastar-announces-windows-based-pbx-system/#comments</comments>
		<pubDate>Fri, 11 Jul 2008 15:23:47 +0000</pubDate>
		<dc:creator>admin</dc:creator>
		
		<category><![CDATA[News]]></category>

		<category><![CDATA[PBX]]></category>

		<category><![CDATA[Skype]]></category>

		<category><![CDATA[windows]]></category>

		<category><![CDATA[Yeastar]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=87</guid>
		<description><![CDATA[BizPBX is Windows-based software PBX solution, which is the world first communication system that realizes the free connection among three different networks Skype, PSTN and SIP. BizPBX business class system not only includes complete and rich functions of PBX, but also contains Instant Messaging and E-Mail system for company internal communication. The simplicity and easy-to-use [...]]]></description>
			<content:encoded><![CDATA[<p><span class="font2">BizPBX is Windows-based software PBX solution, which is the world first communication system that realizes the free connection among three different networks Skype, PSTN and SIP. BizPBX business class system not only includes complete and rich functions of PBX, but also contains Instant Messaging and E-Mail system for company internal communication. The simplicity and easy-to-use of this BizPBX solution makes it especially suitable for small and medium sized company.</span></p>
<table border="0" cellspacing="0" cellpadding="0" width="99%" align="center">
<tbody>
<tr>
<td width="71%">
<p class="font2">1. Support three kinds of trunks,PSTN,SIP and Skype.<br />
2. Support three kinds of extensions,Analog Phone,SIP Phone, Softphone.<br />
3. BizPBX Client software can act as Softphone.<br />
4. BizPBX Client software accessible for every extension.<br />
5. Contacts Listing.Internal Members Listing and displays status of every member.<br />
6. Contacts Listing is sharable among members.<br />
7. Internal E-Mail &amp; Internal Instant Messenger.<br />
8. Voice Mail:Let caller leaves voice message when you are busy or away.<br />
9. Remote Office: Work BizPBX Client software as an extension when working in remote office.<br />
10.Caller ID Profile:Show caller&#8217;s information in real-time.<br />
11.Click to Call:Click on &#8216;name&#8217; to call and save precious time on dialing.<br />
12.Conference Call: Both sides are able to invite others into conference.<br />
13.Free Interoffice Trunking: Make free calls among multiple branches through the interoffice trunking.<br />
14.Web Call: Receive concurrent calls from website with One Skype ID.<br />
15.Virtual Office: Apply local numbers in different countries to reach BizPBX.<span class="style15"><br />
</span></p>
</td>
<td width="29%" valign="top">
<div><img src="http://www.yeastar.com/images/callmanager.gif" alt="" width="216" height="311" /></div>
<div><strong>Call Manager</strong></div>
</td>
</tr>
<tr>
<td colspan="2" height="61"><span class="font2"> <span class="style15">Features:</span></span></td>
</tr>
</tbody>
</table>
<table border="0" cellspacing="0" cellpadding="0" width="100%">
<tbody>
<tr>
<td class="font2" width="38%">Authentication<br />
Automated Attendant<br />
Blind Transfer<br />
Call Booking<br />
Call Detail Records<br />
Call Forward<br />
Call Monitoring<br />
Call Parking<br />
Call Queuing<br />
Call Recording<br />
Call Restriction<br />
Call Retrieval<br />
Call Routing<br />
Call Transfer<br />
Call Waiting<br />
Call Remind</td>
<td class="font2" width="62%">CallerID<br />
Database Integration<br />
Dial by Name<br />
Direct Inward System Access(DISA)<br />
Direct Line<br />
Do Not Disturb<br />
Extension Group<br />
Interactive Voice Response(IVR)<br />
Local and Remote Call Agents<br />
Music On Hold<br />
Protocol Conversion<br />
Remote Call Pickup<br />
Remote Office Support<br />
Roaming Extensions<br />
Share Contacts</td>
</tr>
</tbody>
</table>
]]></content:encoded>
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		</item>
		<item>
		<title>Sangoma Agrees to Acquire Paraxip Technologies</title>
		<link>http://www.voipspeak.net/2008/sangoma-agrees-to-acquire-paraxip-technologies/</link>
		<comments>http://www.voipspeak.net/2008/sangoma-agrees-to-acquire-paraxip-technologies/#comments</comments>
		<pubDate>Mon, 07 Jul 2008 14:55:39 +0000</pubDate>
		<dc:creator>admin</dc:creator>
		
		<category><![CDATA[News]]></category>

		<category><![CDATA[open source]]></category>

		<category><![CDATA[paraxip]]></category>

		<category><![CDATA[Sangoma]]></category>

		<category><![CDATA[VOIP]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=86</guid>
		<description><![CDATA[Company gains foothold in Windows Unified Communications and IP Contact Center Markets
TORONTO, ONTARIO&#8211;(Marketwire - July 7, 2008) - Sangoma® Technologies Corporation (TSXV: STC), the premium provider of PC-based hardware and software for proprietary and open source networking and telephony solutions, today announced that it has entered into an agreement to acquire all of the shares [...]]]></description>
			<content:encoded><![CDATA[<p><em><span style="font-family: Arial,Helvetica,sans-serif; font-size: small;">Company gains foothold in Windows Unified Communications and IP Contact Center Markets</span></em></p>
<p><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;">TORONTO, ONTARIO&#8211;(Marketwire - July 7, 2008) - </span><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;">Sangoma® Technologies Corporation (TSXV: STC), the premium provider of PC-based hardware and software for proprietary and open source networking and telephony solutions, today announced that it has entered into an agreement to acquire all of the shares of Paraxip Technologies Inc. for $4.8 million, payable as to $1.9 million in cash and 2.3 million Sangoma common shares. Paraxip is a leading developer of IP connectivity software that empowers the deployment of IP Telephony applications. Founded in 2003, Paraxip&#8217;s customers include IBM, Genesys (a subsidiary of Alcatel-Lucent), First Data Corporation and the State of California. </span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;">The integration of Paraxip&#8217;s SIP based NetBorder™ suite into Sangoma&#8217;s product line provides Sangoma with a growth path that includes the Unified Communications market, IP contact centers, commercial interactive voice response solutions and other SIP-based telephony markets. The combined products and channels of the two companies considerably extend Sangoma&#8217;s current addressable market range by providing support for all software-based telephony applications.</span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;">&#8220;This acquisition is part of Sangoma&#8217;s strategy for addressing the expanding market for telephony applications of all types, particularly the segment of the market that demands proprietary, well supported solutions,&#8221; said David Mandelstam, President and CEO of Sangoma Technologies. &#8220;The Paraxip transaction not only bolsters Sangoma&#8217;s Windows-based offerings, but also represents a major advancement for the Open Source community. By combining Paraxips gateway products with our premium hardware and drivers, we will provide the Open Source telephony community with a new level of compliance, reliability and scalability.&#8221; </span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;">&#8220;The acquisition is a natural fit that opens up a sizeable Open Source market for Paraxip&#8217;s gateway products and lets us leverage Sangoma&#8217;s worldwide channel of distributors,&#8221; said Serge Forest, Paraxip&#8217;s President and CEO. &#8220;The deal provides Sangoma, in return, with highly differentiated connectivity solutions, including our patent-pending Call Progress Analysis technology, for servicing the fast-growing Contact Center market, as well as access to that market through Paraxip partners. We believe this to be a very synergistic combination.&#8221;</span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;">In addition to at least four other nominees designated by the current Board of Sangoma, Paraxip shareholders will be entitled to nominate two individuals to the Sangoma Board at Sangoma&#8217;s next annual shareholders meeting. The transaction is expected to close on or about July 14, 2008. </span><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;"></p>
<p><strong>About Sangoma Technologies Corporation</strong><br />
Sangoma Technologies Corporation is the premium provider of PC-based hardware and software for proprietary and open source data and telephony transport solutions. The company develops and manufactures the most scalable and reliable voice and Wide Area Network data cards in the industry, including the award-winning Advanced Flexible Telecommunications (AFT) product line. Founded in 1984, Sangoma Technologies Corporation is publicly traded on the TSX Venture Exchange (TSXV: STC - News). Additional information on Sangoma can be found at: <a href="http://www.sangoma.com/" target="_blank">www.sangoma.com</a>.</span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;"><em>The TSX Venture Exchange does not accept responsibility for the adequacy or accuracy of this release.</em></span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;"><strong>About Paraxip Technologies </strong><br />
Paraxip develops and markets connectivity products that empower the rapid and cost effective deployment of IP Telephony in the contact center and in the enterprise. Paraxip&#8217;s flagship software product, NetBorder, leverages standard computing platforms to provide smart connectivity to sophisticated telephony applications. The patent-pending NetBorder architecture reduces deployment costs and allows IP Telephony solutions in the enterprise to seamlessly extend their reach to numerous external devices and networks. Additional information on Paraxip can be found at: <a href="http://www.paraxip.com/" target="_blank">www.paraxip.com</a>.</span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;">Sangoma is a registered trademark of Sangoma Technologies Corporation. NetBorder is a trademark of Paraxip Technologies. All other trademarks are the property of their respective owners.</span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;"><strong>Caution Concerning Forward-Looking Statements </strong><br />
This news release contains forward-looking statements relating to the proposed acquisition of Paraxip Technologies Inc., including statements regarding the completion of the proposed transaction and other statements that are not historical facts. Such forward-looking statements are subject to important risks, uncertainties and assumptions. The results or events predicted in these forward-looking statements may differ materially from actual results or events. As a result, you are cautioned not to place undue reliance on these forward-looking statements.</span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;">The completion of the proposed transaction is subject to a number of conditions which may not be satisfied in accordance with their terms, and/or the parties to the share purchase agreement may exercise their termination rights, in which case the proposed transaction could be modified, restructured or terminated, as applicable.</span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;">The forward-looking statements in this news release are made as of the date of this release. We undertake no obligation to comment on expectations of, or statements made by third parties in respect of the proposed transaction.</span><span style="font-family: Arial,Helvetica,sans-serif;"><span style="font-size: x-small;"></p>
<p></span></span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;"> Media Contacts:<br />
Kathleen Reed<br />
Marketing Director, Sangoma Technologies Corporation<br />
(905) 474-1990 ext 115<br />
Email: <a href="mailto:kathleen@sangoma.com" target="_blank">kathleen@sangoma.com</a><br />
Website: <a href="http://cmpgnr.com/r.html?c=1265260&amp;r=1264176&amp;t=1273426768&amp;l=1&amp;d=89813620&amp;u=http%3a%2f%2fwww%2esangoma%2ecom&amp;g=0&amp;f=-1" target="_blank">www.sangoma.com</a></p>
<p>or</p>
<p>MRB Public Relations (For Sangoma)<br />
Michael Becce<br />
(732) 758-1100 Ext. 104<br />
Email: <a href="mailto:trichardson@mrb-pr.com" target="_blank">mbecce@mrb-pr.com<br />
</a></span></p>
<p align="left"><span style="font-family: Arial,Helvetica,sans-serif; font-size: x-small;">or</p>
<p>Frederic Dickey<br />
Market Development Director<br />
Paraxip Technologies Inc.<br />
(514) 282-7111, ext. 233<br />
Email: <a href="mailto:frederic.dickey@paraxip.com" target="_blank">frederic.dickey@paraxip.com</a></span></p>
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		<title>Polycom Desktop IP Phone Combines Color Display, HD Voice, Gigabit Ethernet, And Industry&#8217;s First Color Attendant Console</title>
		<link>http://www.voipspeak.net/2008/polycom-desktop-ip-phone-combines-color-display-hd-voice-gigabit-ethernet-and-industrys-first-color-attendant-console/</link>
		<comments>http://www.voipspeak.net/2008/polycom-desktop-ip-phone-combines-color-display-hd-voice-gigabit-ethernet-and-industrys-first-color-attendant-console/#comments</comments>
		<pubDate>Wed, 25 Jun 2008 13:18:57 +0000</pubDate>
		<dc:creator>admin</dc:creator>
		
		<category><![CDATA[Featured Articles]]></category>

		<category><![CDATA[announcement]]></category>

		<category><![CDATA[color]]></category>

		<category><![CDATA[LCD]]></category>

		<category><![CDATA[Polycom]]></category>

		<category><![CDATA[SIP]]></category>

		<category><![CDATA[soundpoint]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=84</guid>
		<description><![CDATA[SoundPoint IP 670 Delivers a Rich Voice and Visual Experience to Increase the Speed of Business
INFOCOMM, LAS VEGAS and PLEASANTON, Calif., June 18, 2008 – Polycom, Inc. (NASDAQ: PLCM), the leader in unified collaborative communications solutions, today announced the newest addition to its SoundPoint® IP desktop phone line, the SoundPoint IP 670. The SoundPoint IP [...]]]></description>
			<content:encoded><![CDATA[<p align="left"><strong><em>SoundPoint IP 670 Delivers a Rich Voice and Visual Experience to Increase the Speed of Business</em></strong></p>
<p><img class="alignright" style="float: right;" src="http://polycom.com/usa/en/images/products/voice/desktop/soundpoint_ip/soundpoint_ip_670.jpg" alt="" width="200" height="200" /><strong>INFOCOMM, LAS VEGAS and PLEASANTON, Calif., June 18, 2008</strong> – Polycom, Inc. (NASDAQ: PLCM), the leader in unified collaborative communications solutions, today announced the newest addition to its SoundPoint® IP desktop phone line, the SoundPoint IP 670. The SoundPoint IP 670 is an application-enabled desktop SIP phone with a high-performance color display, Polycom&#8217;s revolutionary HD Voice technology, and Gigabit Ethernet (GigE) connectivity. It provides professionals with an intuitive color interface for easier viewing and navigation of phone functions and productivity-enhancing applications.</p>
<p>The SoundPoint IP 670 also features the industry&#8217;s first color expansion module. When equipped with up to three color expansion modules, the SoundPoint IP 670 delivers the industry&#8217;s first color attendant console solution for call attendants and administrative assistants. This solution significantly increases the call handling capability of the phone and enables attendants to better manage incoming calls by being able to view presence status without having to be in front of a PC.</p>
<p>&#8220;Polycom&#8217;s new SoundPoint IP 670 phone is a good choice for organizations that are pursuing a standards-based communication strategy and are planning to embed applications on the phone,&#8221; said Matthias Machowinski, directing analyst with Infonetics Research. &#8220;And what&#8217;s noteworthy for power users is that it operates at 1 Gbps, so it doesn&#8217;t introduce a new congestion point on the network.&#8221;</p>
<p>&#8220;The SoundPoint IP 670 provides a full-color user interface to deliver a significantly augmented and visually pleasing user experience when running productivity-enhancing applications, such as the Polycom Productivity Suite,&#8221; said Victor Yue, Director, Fujitsu Asia Pte Ltd, &#8220;We applaud Polycom for enabling us to continue to offer our customers the most comprehensive, best-sounding, most interoperable, high-quality IP telephony solutions for small and medium–sized businesses and enterprises.&#8221;</p>
<p>The SoundPoint IP 670 supports six-lines and includes advanced SIP features and capabilities, such as support for shared lines, text messaging, and buddy presence monitoring. It features 14 default color background displays for phone personalization. Additional customized backgrounds, such as a company logo, can be added to deliver a rich desktop branding experience. The phone also includes an integrated XHTML micro-browser that enables users to take advantage of productivity-enhancing Web-based applications and also provides the ideal platform for Polycom&#8217;s recently announced Polycom Productivity Suite for SoundPoint IP phones.</p>
<p>As the industry&#8217;s first color expansion module, the SoundPoint IP Color Expansion Module augments the color user interface of the SoundPoint IP 670 phone.  It features 14 multifunctional line keys that can be set up as line registration, call appearance, speed-dial, or a direct station select (DSS)/busy lamp field (BLF) keys. Up to three Color Expansion Modules can be snapped onto the SoundPoint IP 670 to form a full-featured color call attendant console solution. The solution supports up to 34 line registrations, 47 BLF monitored lines, and 24 concurrent calls. This enables the telephone attendant to promptly accept, screen, dispatch and effortlessly monitor incoming calls.<strong> </strong></p>
<p>&#8220;We&#8217;re always seeking ways to improve the customer experience and offer productivity-enhancing features that drive the speed of business,&#8221; said Sunil Bhalla, senior vice president and general manager of Polycom&#8217;s Voice Communications Solutions division. &#8220;The SoundPoint IP 670 combines a rich color display with Polycom HD Voice to offer our customers a visually pleasing user interface, an incredible sounding voice experience and a future-proof platform with Gigabit Ethernet.&#8221;</p>
<p>The SoundPoint IP phone is being certified to deliver comprehensive interoperability and extensive feature support with Polycom&#8217;s growing list of more than 24 SIP-based call control platform partners including 3Com, BroadSoft, Digium, Interactive Intelligence, Sylantro, and other Polycom VoIP Interoperability Partners (VIP). For more information on the Polycom VIP Program, visit <a href="http://www.polycom.com/vip" target="_blank">www.polycom.com/vip</a><strong>.</strong></p>
<p><strong>Pricing and Availability</strong><br />
The Power over Ethernet (PoE) version of the Polycom SoundPoint IP 670 is available in North America, most of Central and Latin America, Europe and most of Asia-Pac today through Polycom certified channel partners for an MSRP of U.S. $599.   The SoundPoint IP Color Expansion Modules are available for an MSRP of $319 each.  To learn more about the SoundPoint IP 670 and its Color Expansion Modules, please visit the Polycom web site at <a href="http://www.polycom.com/voip" target="_blank">www.polycom.com/voip</a>.</p>
<p><strong>About Polycom</strong><br />
Polycom, Inc. is the worldwide leader in unified collaborative communications (UCC) that maximize the efficiency and productivity of people and organizations. Polycom delivers the broadest array of high definition telepresence video, wired and wireless voice, and content solutions so people can enjoy the best communications, whether from a real-time collaborative interaction or on-demand streamed video experience. Spanning from the desktop to the industry&#8217;s only immersive telepresence suite, Polycom&#8217;s high quality collaboration and communications solutions are easy to deploy and manage, as well as intuitive to use. Based on open standards, they integrate seamlessly with leading telephony and presence-based networks. With innovative market-driving technologies, a compelling vision for next generation visual communications, best-in-class products, alliance partnerships, and world-class service, Polycom is the smart choice for organizations to gain a competitive advantage with proven and trusted communication solutions.  For additional information, call 800-POLYCOM or visit the Polycom web site at <a href="http://www.polycom.com/" target="_blank">www.polycom.com</a>.</p>
<p>Polycom reserves the right to modify future product plans at any time. Products and/or related specifications referenced in this press release are not guaranteed, and will be delivered on a when and if available basis.</p>
<p>Polycom, the Polycom logo, and SoundPoint are registered trademarks, and Polycom HD Voice technology is a trademark of Polycom in the U.S. and various countries. All other trademarks are the property of their respective owners. ©2008, Polycom, Inc. All rights reserved.</p>
<table border="0">
<tbody>
<tr>
<td valign="top">Contact:</td>
<td>Robin Raulf-Sager<br />
Polycom, Inc.<br />
303.583.5342<br />
<a href="mailto:robin.raulf-sager@polycom.com">robin.raulf-sager@polycom.com</a></td>
</tr>
</tbody>
</table>
]]></content:encoded>
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		<title>3CX Develops 10 Key New Features for Its IP PBX in Only 20 Weeks</title>
		<link>http://www.voipspeak.net/2008/3cx-develops-10-key-new-features-for-its-ip-pbx-in-only-20-weeks/</link>
		<comments>http://www.voipspeak.net/2008/3cx-develops-10-key-new-features-for-its-ip-pbx-in-only-20-weeks/#comments</comments>
		<pubDate>Tue, 24 Jun 2008 13:52:24 +0000</pubDate>
		<dc:creator>admin</dc:creator>
		
		<category><![CDATA[News]]></category>

		<category><![CDATA[3cx]]></category>

		<category><![CDATA[IP PBX]]></category>

		<category><![CDATA[SIP]]></category>

		<category><![CDATA[VOIP]]></category>

		<category><![CDATA[windows]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=81</guid>
		<description><![CDATA[Call conferencing, Paging and Windows 2008 support among major features added to software-based 3CX Phone System for Windows in record time
London, UK – 24 June 2008 - 3CX (www.3cx.com) has announced the release of version 6.0 of 3CX Phone System for Windows. The most recent version of the award-winning and software-based PBX ships with 10 [...]]]></description>
			<content:encoded><![CDATA[<p><strong>Call conferencing, Paging and Windows 2008 support among major features added to software-based 3CX Phone System for Windows in record time</strong></p>
<p><a href="http://www.voipspeak.net/wp-content/uploads/2008/06/3cx1.gif" rel="thumbnail"><img class="alignnone size-full wp-image-83 alignright" style="float: right;" title="3cx1" src="http://www.voipspeak.net/wp-content/uploads/2008/06/3cx1.gif" alt="" width="100" height="39" /></a>London, UK – 24 June 2008 - 3CX (www.3cx.com) has announced the release of version 6.0 of 3CX Phone System for Windows. The most recent version of the award-winning and software-based PBX ships with 10 key new enterprise-level features such as call conferencing, paging and Windows 2008 support. These 10 new features were developed by the international software development company in only 20 weeks.</p>
<p>“We are very excited to launch this new version of 3CX Phone System for Windows. 3CX’ development team has proved that we can turn around new versions and new features in record time. What would take years on traditional PBXs takes only months on a software-based PBX like ours; simply because the structure is much more manageable” said Nick Galea, CEO at 3CX.</p>
<p>“With our new Windows 2008 support, organizations can virtualize the PBX and run a phone system without the need for an appliance or an extra server. This means lower administration costs, no extra hardware and less energy consumption.”</p>
<p><strong>10 Key New Features of 3CX Phone System for Windows v6.0</strong></p>
<ol>
<li>Call conference service – up to 32 participants.</li>
<li>Windows 2008 server support – run a PBX virtualized.</li>
<li>Paging – page groups of users to broadcast a message.</li>
<li>Intercom – with two-way audio capability.</li>
<li>Support for Vegastream gateways (Patton &amp; Audiocode already supported).</li>
<li>Enhanced SIP interoperability – addition of support for many leading international VoIP providers and SIP phones.</li>
<li>BLF provisioning – automatic provision of BLF lights indicating extension status on phones.</li>
<li>Phonebook Provisioning – listing all extensions and allowing addition of custom entries.</li>
<li>Fax – 3CX’ fax server allows the whole network to send and receive faxes.</li>
<li>Extended HTTP API - ability to switch recording on/off, disable an extension, and disable outbound calls for an extension.</li>
</ol>
<p><a href="http://www.3cx.com/phone-system/" target="_blank">3CX Phone System for Windows</a> allows businesses to completely break free from the restrictions of costly hardware-based, proprietary phone systems. It also includes many other features that make 3CX Phone System easy to install, configure, manage and use; helping businesses increase employees’ mobility and productivity.</p>
<p><strong>Four available editions: Small Business, Pro, Enterprise and Free</strong></p>
<p>3CX Phone System is available in four editions, all supporting an unlimited number of extensions. The Free edition is limited to 8 simultaneous calls, whereas, the commercial editions can be extended beyond that and have a more extensive feature set. The Small Business version supports up to 8, the Pro version up to 16 and the Enterprise edition supports up to 32 simultaneous calls. Call capacity can be expanded with upgrade packs. A detailed edition comparison chart can be found on <a href="http://www.3cx.com/phone-system/edition-comparison.html" target="_blank">http://www.3cx.com/phone-system/edition-comparison.html</a>.</p>
<p><strong>3CX Phone System Free edition</strong></p>
<p>A Free edition, supporting an unlimited number of extensions, is also available. It can be downloaded from <a href="http://www.3cx.com/phone-system/download-phone-system.html" target="_blank">http://www.3cx.com/phone-system/download-phone-system.html</a>.</p>
<p><strong>About 3CX</strong></p>
<p>3CX is an international developer of telecommunications software, headquartered in Europe with offices in the UK, USA, Germany, Cyprus, Malta and Hong Kong. It is a Microsoft Gold Certified partner and is backed by an experienced management and development team. Its product, 3CX Phone System for Windows, developed specifically for the SMB (small &amp; medium business) market, has earned Windows Server 2003 Certification and has received numerous awards, including the TMC Labs 2007 Innovation Award, The Windowsnetworking.com Gold Award, as well as, the IT EXPO Best of Show award 2007, the INTERNET Telephony Magazine Product of the Year Award, and the Communications Solutions 2007 Award, all in recognition to the company’s commitment to innovation and quality. 3CX maintains a global presence with offices in six countries and localized information available in various languages. For more information, visit <a href="http://www.3cx.com/">www.3cx.com</a>.</p>
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		<title>OpenVox® New Product B100M Mini-PCI ISDN BRI Card Is Released</title>
		<link>http://www.voipspeak.net/2008/openvox%c2%ae-new-product-b100m-mini-pci-isdn-bri-card-is-released/</link>
		<comments>http://www.voipspeak.net/2008/openvox%c2%ae-new-product-b100m-mini-pci-isdn-bri-card-is-released/#comments</comments>
		<pubDate>Tue, 24 Jun 2008 13:17:14 +0000</pubDate>
		<dc:creator>admin</dc:creator>
		
		<category><![CDATA[Featured Articles]]></category>

		<category><![CDATA[News]]></category>

		<category><![CDATA[Asterisk]]></category>

		<category><![CDATA[BRI]]></category>

		<category><![CDATA[elastix]]></category>

		<category><![CDATA[ISDN]]></category>

		<category><![CDATA[OpenVox]]></category>

		<category><![CDATA[trixbox]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=80</guid>
		<description><![CDATA[SHENZHEN, CHINA&#8211;OpenVox®, the largest asterisk® telephony hardware manufacturer in China, has now released the new product B100M. OpenVox B100M is a Mini PCI type III BRI card supporting one BRI S/T interface. NT/TE mode can be independently configured on the port. B100M can be used for building open source asterisk® based systems such as ISDN [...]]]></description>
			<content:encoded><![CDATA[<p>SHENZHEN, CHINA&#8211;OpenVox®, the largest asterisk® telephony hardware manufacturer in China, has now released the new product B100M. OpenVox B100M is a Mini PCI type III BRI card supporting one BRI S/T interface. NT/TE mode can be independently configured on the port. B100M can be used for building open source asterisk® based systems such as ISDN PBX and VoIP gateway.</p>
<p><strong><span style="font-family: Arial; font-size: small;"><span style="font-size: 12pt; font-weight: bold; font-family: Arial;">Target Applications:</span></span></strong></p>
<ul>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">High performance ISDN PC cards</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">ISDN PABX for BRI</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">VoIP gateways</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">ISDN LAN routers for BRI</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">ISDN least cost routers for BRI</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">ISDN test equipment for BRI</span></span></li>
</ul>
<p><strong><span style="font-family: Arial; font-size: small;"><span style="font-size: 12pt; font-weight: bold; font-family: Arial;">Features:</span></span></strong></p>
<ul>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">One integrated S/T interface</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">ITU-T I.430 and TBR 3 certified and S/T ISDN supporting in TE and NT mode</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">DTMF detection on all B-channels</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">RoHS compliant</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">Multiparty audio conferences bridge</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">Support Mini PCI type III</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">Designed for low-power systems</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">Support AskoziaPBX system, trixbox</span></span><sup><span style="font-family: Arial; font-size: small;"><span style="font-size: 12pt; font-family: Arial;">®</span></span></sup><span style="font-family: Arial;"><span style="font-family: Arial;">, Elastix</span></span><sup><span style="font-family: Arial; font-size: small;"><span style="font-size: 12pt; font-family: Arial;">®</span></span></sup><span style="font-family: Arial;"><span style="font-family: Arial;"> and other asterisk</span></span><sup><span style="font-family: Arial; font-size: small;"><span style="font-size: 12pt; font-family: Arial;">®</span></span></sup><span style="font-family: Arial; font-size: x-small;"><span style="font-family: Arial;"> based distributions</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">Support VIA, PC Engines motherboard and AMD geode based motherboard</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">The port can be configured for TE or NT mode</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">Support Bristuff, ISDN4BSD and mISDN driver</span></span></li>
<li><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">Application ready: Use asterisk</span></span><sup><span style="font-family: Arial; font-size: small;"><span style="font-size: 12pt; font-family: Arial;">®</span></span></sup><span style="font-family: Arial;"><span style="font-family: Arial;"> to build your IP-PBX/Voicemail system</span></span></li>
</ul>
<p><strong><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-weight: bold; font-family: Arial;">About OpenVox Communication Co., Ltd</span></span></strong><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;"> </span></span></p>
<p><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">OpenVox Communication Co., Ltd is dedicated its passion in supplying Open Source Computer Telephony hardware and software products. With the people’s expertise in design and service experience, we provide professional quality products and a 3-month “no questions asked” return policy for all OpenVox</span></span><sup><span style="font-family: Arial; font-size: small;"><span style="font-size: 12pt; font-family: Arial;">®</span></span></sup><span style="font-family: Arial;"><span style="font-family: Arial;"> hardware.</span></span></p>
<p><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">Asterisk is a registered trademark of Digium, Inc. </span></span></p>
<p><span style="font-family: Arial; font-size: x-small;"><span style="font-size: 10.5pt; font-family: Arial;">OpenVox Business Contact:<br />
Phone: +86-755-82535461   +86-755-82535462   +86-755-82535362<br />
Email: <a href="mailto:sales@openvox.com.cn" target="_blank">sales@openvox.com.cn<br />
</a>Website: <a href="http://www.openvox.com.cn/" target="_blank">www.openvox.com.cn</a> </span></span></p>
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		<title>Asternic Releases New Call Center Stats</title>
		<link>http://www.voipspeak.net/2008/asternic-releases-new-call-center-stats/</link>
		<comments>http://www.voipspeak.net/2008/asternic-releases-new-call-center-stats/#comments</comments>
		<pubDate>Thu, 19 Jun 2008 05:17:45 +0000</pubDate>
		<dc:creator>admin</dc:creator>
		
		<category><![CDATA[News]]></category>

		<category><![CDATA[Asterisk]]></category>

		<category><![CDATA[cdr]]></category>

		<category><![CDATA[queue]]></category>

		<category><![CDATA[reports]]></category>

		<category><![CDATA[stats]]></category>

		<category><![CDATA[trixbox]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=77</guid>
		<description><![CDATA[Asternic, the creator of the popular Flash Operator Panel, has released a new Call Center Stats package.
Asternic Call Center Stats is a package to analyze your Asterisk PBX queue_log files and display asterisk queue information in real time on a web page.
There are several reports with nice flash graphics and option to export to pdf [...]]]></description>
			<content:encoded><![CDATA[<p><a href="http://www.voipspeak.net/wp-content/uploads/2008/06/callstats.gif" rel="thumbnail"><img class="alignnone size-medium wp-image-78 alignright" style="float: right;" title="callstats" src="http://www.voipspeak.net/wp-content/uploads/2008/06/callstats-300x201.gif" alt="" width="300" height="201" /></a>Asternic, the creator of the popular Flash Operator Panel, has released a new Call Center Stats package.</p>
<p>Asternic Call Center Stats is a package to analyze your Asterisk PBX queue_log files and display asterisk queue information in real time on a web page.</p>
<p>There are several reports with nice flash graphics and option to export to pdf and csv (Excel).</p>
<p>There are two versions, a free GPLv3 licenced and a commercial one</p>
<p>The commercial version includes several more reports, user levels and access, realtime parsing of queue_log, ability to listen to call recordings via streaming, etc. I can provide access to an online demo upon request.</p>
<p>Source: <a href="http://www.asternic.org/stats/" target="_blank">Asternic</a></p>
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		<title>trixbox Live! Sells Out, Event Expanded, Registration Reopened for Limited Time</title>
		<link>http://www.voipspeak.net/2008/trixbox-live-sells-out-event-expanded-registration-reopened-for-limited-time/</link>
		<comments>http://www.voipspeak.net/2008/trixbox-live-sells-out-event-expanded-registration-reopened-for-limited-time/#comments</comments>
		<pubDate>Tue, 17 Jun 2008 05:35:18 +0000</pubDate>
		<dc:creator>admin</dc:creator>
		
		<category><![CDATA[Featured Articles]]></category>

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		<description><![CDATA[Free Gathering for trixbox Community Happens July 15th in Los Angeles
LOS ANGELES&#8211;(BUSINESS WIRE)&#8211;Fonality®, the leading provider of open source        business communications systems, announced today that it is has expanded        the July 15th trixbox Live! event, a free     [...]]]></description>
			<content:encoded><![CDATA[<p><em>Free Gathering for trixbox Community Happens July 15<sup id="bwanpa12">th</sup> in Los Angeles</em></p>
<p>LOS ANGELES&#8211;(<a href="http://www.businesswire.com/">BUSINESS WIRE</a>)&#8211;Fonality<span id="bwanpa2">®</span>, the leading provider of open source        business communications systems, announced today that it is has expanded        the July 15<sup id="bwanpa11">th</sup> trixbox Live! event, a free        gathering for the trixbox community being held at the Westin LAX in Los        Angeles. trixbox Live! sold out on the first day of registration and has        now been expanded to accommodate all trixbox community members. Designed        as a free networking and education event for the largest open source        telephony community, attendees are coming from around the United States        and 20 different countries. To reserve space, register today at <a href="http://trixboxlive.eventbrite.com/" target="_blank">http://trixboxlive.eventbrite.com</a>.</p>
<p>trixbox Live! is the first in a series of free nationwide events created        especially for the trixbox CE and trixbox Pro communities. trixbox        founder, Andrew Gillis, along with trixbox Community Director, Kerry        Garrison, will host a full day of training on how to take trixbox to        your market and be successful selling trixbox-based phone systems.</p>
<p><span id="bwanpa3">“</span>I am humbled and excited by growth of the        trixbox project,<span id="bwanpa4">”</span> says Andrew Gillis, trixbox        founder. <span id="bwanpa5">“</span>It started as a simple means of        installing Asterisk for myself and some friends and has grown into the        largest project in the world for open source telephony. And now with the        trixbox Live! events, we can bring the community face to face with each        other and spur more innovation and a deeper sense of community.<span id="bwanpa6">”</span></p>
<p>trixbox Live! attendees will learn about new features, get insight into        the trixbox roadmap and see live demos that demonstrate how to integrate        trixbox with a variety of different hardware components.        trixbox-specific sales training will include:</p>
<ul>
<li class="bwlistitemmarginbottom"> How to explain the value of trixbox</li>
<li class="bwlistitemmarginbottom"> How to effectively sell against competing products</li>
<li class="bwlistitemmarginbottom"> Details on effective closing techniques</li>
<li class="bwlistitemmarginbottom"> How to increase your profits by cross-selling services and third-party          products</li>
</ul>
<p>All attendees will receive a free trixbox Pro demo kit, lots of great        trixbox schwag and a free community networking lunch. The demo kit        includes two trixbox Pro 2.0 Call Center Edition lifetime licenses with        one month of unlimited technical support. As part of the sales training        agenda, Brad Pitt, director of Fonality channel sales, will provide an        overview of trixbox Pro 2.0 with HUD, the award-winning application that        provides a unified view of chat, email, and calling for all fixed and        mobile workers. For more information about trixbox Live! Visit <a href="http://trixbox.org/event/trixboxlive" target="_blank">http://trixbox.org/event/trixboxlive</a>.</p>
<p>trixbox CE is the world<span id="bwanpa7">’</span>s largest free and        open source telephony project with 3 million downloads, 200,000 live        deployments, and is growing at 125,000 new downloads each month. Just        released, trixbox CE 2.6.1 has new, expanded hardware support, an        enhanced user interface and new admin modules, including an improved        backup system and a bulk extension import tool.</p>
<p>trixbox Pro, a commercial offering, is designed for modern workplaces        where companies have a mix of office, roaming and home-based workers. It        supports both VoIP calling and traditional phone lines, and a patented        hybrid-hosted architecture allows employee identity to be maintained        whether they are working at the office, at home or traveling.</p>
<p><strong>About Fonality</strong></p>
<p>Fonality, <a href="http://www.fonality.com/" target="_blank">www.fonality.com</a>, is a        leader in business phone systems and contact center solutions for small        and medium-sized businesses. Used by over 5,000 companies and 100,000        end users in 100 countries, Fonality&#8217;s award winning IP-PBX VoIP phone        systems have connected more than 225,000,000 mission critical phone        calls. The PBXtra and trixbox Pro product lines are based on Fonality&#8217;s        patent-pending Anywhere Management<span id="bwanpa8">™</span> Hybrid-Hosted<span id="bwanpa9">™</span> architecture, plus an improved        version of the popular open source Asterisk code base which has been        modified to add reliability, stability and enterprise-class features.        PBXtra and trixbox Pro deliver the advanced capabilities of an        enterprise-class phone system for 40 to 80 percent less than traditional        offerings. Fonality&#8217;s fully free and open source telephony platform,        trixbox CE (<a href="http://www.trixbox.org/" target="_blank">www.trixbox.org</a>), is        home to one of the world&#8217;s largest and fastest growing communities of        open source telephony users, with over 200,000 live deployments and        125,000 new downloads each month. Fonality headquarters are in Los        Angeles and company investors include Azure Capital Partners and Intel        Capital.</p>
<p>Fonality, PBXtra, trixbox, and HUD are registered trademarks and        Hybrid-Hosted and Anywhere Management are trademarks of Fonality. Other        trademarks are the property of their respective holders.</p>
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