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	<title>VoipSpeak &#187; Digium</title>
	<atom:link href="http://voipspeak.net/tag/digium/feed/" rel="self" type="application/rss+xml" />
	<link>http://voipspeak.net</link>
	<description>Just another VoIP blog</description>
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		<title>Configuring Digium Analog Cards with trixbox CE</title>
		<link>http://voipspeak.net/2010/03/configuring-digium-tdm410ptdm800p-analog-cards-with-trixbox-ce/</link>
		<comments>http://voipspeak.net/2010/03/configuring-digium-tdm410ptdm800p-analog-cards-with-trixbox-ce/#comments</comments>
		<pubDate>Tue, 16 Mar 2010 15:06:00 +0000</pubDate>
		<dc:creator>KerryG</dc:creator>
				<category><![CDATA[Tutorials]]></category>
		<category><![CDATA[Analog]]></category>
		<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Digium]]></category>
		<category><![CDATA[FXO]]></category>
		<category><![CDATA[FXS]]></category>
		<category><![CDATA[trixbox CE]]></category>

		<guid isPermaLink="false">http://voipspeak.net/?p=291</guid>
		<description><![CDATA[Many people don&#8217;t realize just how easy it is to configure a Digium analog interface card with trixbox CE. Not that it is very difficult with other distros, its usually only a few simple commands, but trixbox CE makes it even easier with one command that configures the card very easily. The same setup-pstn command [...]]]></description>
			<content:encoded><![CDATA[<p><object id="wistia_90120" classid="clsid:d27cdb6e-ae6d-11cf-96b8-444553540000" width="560" height="315" codebase="http://download.macromedia.com/pub/shockwave/cabs/flash/swflash.cab#version=6,0,40,0"><param name="allowfullscreen" value="true" /><param name="allowscriptaccess" value="always" /><param name="wmode" value="opaque" /><param name="flashvars" value="playButtonVisible=true&amp;unbufferedSeek=true&amp;controlsVisibleOnLoad=false&amp;autoPlay=false&amp;videoUrl=http://embed.wistia.com/deliveries/c083d67717f3179478f64e9b088b03ab3071e8d2.bin&amp;stillUrl=http://embed.wistia.com/deliveries/c6b563383c63ade642b0e2a4695fea341b066801.bin&amp;embedServiceURL=http://distillery.wistia.com/x&amp;accountKey=wistia-production_542&amp;mediaID=wistia-production_90120&amp;mediaDuration=375.1" /><param name="src" value="http://embed.wistia.com/flash/embed_player_v1.1.swf" /><param name="name" value="wistia_90120" /><embed id="wistia_90120" type="application/x-shockwave-flash" width="560" height="315" src="http://embed.wistia.com/flash/embed_player_v1.1.swf" name="wistia_90120" flashvars="playButtonVisible=true&amp;unbufferedSeek=true&amp;controlsVisibleOnLoad=false&amp;autoPlay=false&amp;videoUrl=http://embed.wistia.com/deliveries/c083d67717f3179478f64e9b088b03ab3071e8d2.bin&amp;stillUrl=http://embed.wistia.com/deliveries/c6b563383c63ade642b0e2a4695fea341b066801.bin&amp;embedServiceURL=http://distillery.wistia.com/x&amp;accountKey=wistia-production_542&amp;mediaID=wistia-production_90120&amp;mediaDuration=375.1" wmode="opaque" allowscriptaccess="always" allowfullscreen="true"></embed></object></p>
<p>Many people don&#8217;t realize just how easy it is to configure a Digium analog interface card with trixbox CE. Not that it is very difficult with other distros, its usually only a few simple commands, but trixbox CE makes it even easier with one command that configures the card very easily. The same <em>setup-pstn</em> command will work on all versions of trixbox CE since version 2.4.<span id="more-291"></span>Once you have to card in the system, boot it up, log in as root and type the <em>setup-pstn</em> command. If you see the channels come up at the end then just type <em>amportal restart</em> and Asterisk will restart. Then you need to go into the web interface and make sure your trunks are configured properly. After that, you are ready to make and receive calls.</p>
<h3>VoipStore.com Feature Solutions Partner Links</h3>
<ul>
<li><a href="http://www.voipstore.com/featured-solutions/digium">Digium</a></li>
<li><a href="http://www.888voipstore.com/digium">Digium products on 888VoipStore.com</a></li>
<li><a href="http://www.voipstore.com/featured-solutions/trixbox">Fonality / trixbox</a></li>
<li><a href="http://www.888voipstore.com/trixbox">trixbox products on 888VoipStore.com</a></li>
</ul>
 ]]></content:encoded>
			<wfw:commentRss>http://voipspeak.net/2010/03/configuring-digium-tdm410ptdm800p-analog-cards-with-trixbox-ce/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Digium TDM410P Analog Interface Card Overview</title>
		<link>http://voipspeak.net/2010/03/digium-tdm410p-analog-interface-card-overview/</link>
		<comments>http://voipspeak.net/2010/03/digium-tdm410p-analog-interface-card-overview/#comments</comments>
		<pubDate>Tue, 16 Mar 2010 15:03:32 +0000</pubDate>
		<dc:creator>KerryG</dc:creator>
				<category><![CDATA[Feature]]></category>
		<category><![CDATA[Reviews]]></category>
		<category><![CDATA[Analog]]></category>
		<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Digium]]></category>
		<category><![CDATA[FXO]]></category>
		<category><![CDATA[FXS]]></category>

		<guid isPermaLink="false">http://voipspeak.net/?p=289</guid>
		<description><![CDATA[Kerry Garrison and the team at VoipStore.com have posted another great product overview, this one is on the improved analog cards from Digium. The Digium® TDM410 is a half-length PCI 2.2-compliant modular gateway card for connecting analog telephone stations and analog POTS lines through a PC. It supports a combination of up to four station [...]]]></description>
			<content:encoded><![CDATA[<p><object id="wistia_86859" classid="clsid:d27cdb6e-ae6d-11cf-96b8-444553540000" width="560" height="315" codebase="http://download.macromedia.com/pub/shockwave/cabs/flash/swflash.cab#version=6,0,40,0"><param name="allowfullscreen" value="true" /><param name="allowscriptaccess" value="always" /><param name="wmode" value="opaque" /><param name="flashvars" value="playButtonVisible=true&amp;unbufferedSeek=true&amp;controlsVisibleOnLoad=false&amp;autoPlay=false&amp;videoUrl=http://embed.wistia.com/deliveries/88201ede3e7226be2ea59488ede5c03ea2b1a7af.bin&amp;stillUrl=http://embed.wistia.com/deliveries/3ac4505b215fa0b22365afc350510befd1678b66.bin&amp;embedServiceURL=http://distillery.wistia.com/x&amp;accountKey=wistia-production_542&amp;mediaID=wistia-production_86859&amp;mediaDuration=279" /><param name="src" value="http://embed.wistia.com/flash/embed_player_v1.1.swf" /><embed id="wistia_86859" type="application/x-shockwave-flash" width="560" height="315" src="http://embed.wistia.com/flash/embed_player_v1.1.swf" flashvars="playButtonVisible=true&amp;unbufferedSeek=true&amp;controlsVisibleOnLoad=false&amp;autoPlay=false&amp;videoUrl=http://embed.wistia.com/deliveries/88201ede3e7226be2ea59488ede5c03ea2b1a7af.bin&amp;stillUrl=http://embed.wistia.com/deliveries/3ac4505b215fa0b22365afc350510befd1678b66.bin&amp;embedServiceURL=http://distillery.wistia.com/x&amp;accountKey=wistia-production_542&amp;mediaID=wistia-production_86859&amp;mediaDuration=279" wmode="opaque" allowscriptaccess="always" allowfullscreen="true"></embed></object></p>
<p>Kerry Garrison and the team at <a href="http://www.voipstore.com">VoipStore.com</a> have posted another great product overview, this one is on the improved analog cards from Digium.</p>
<p>The Digium® TDM410 is a half-length PCI 2.2-compliant modular gateway   card for connecting analog telephone stations and analog POTS lines   through a PC. It supports a combination of up to four station or trunk   modules for a total of 4 lines.</p>
<p>Using Digium&#8217;s Asterisk® software and standard PC hardware, one can   create a telephony environment that includes all of the sophisticated   features of a high-end business telephone system.</p>
<p>Using an  industry-standard bursting, bus-mastering interface chip  that is found  within millions of PC systems worldwide, and Digium  VoiceBus™  technology, the TDM410 eliminates the requirement for  external  gateways, with industry-leading performance and price. The  trunk and  station modules are interchangeable, allowing the creation of  any  combination of interfaces. The optional hardware echo cancellation   module provides 1024 taps (128 milliseconds) of echo cancellation for   superior voice quality on both trunk and station interfaces. Scaling of   this solution is accomplished by adding additional TDM410 or other   Digium analog interface cards.</p>
<p>Digium Analog cards, with there interchangeable single and quad FXS  and FXO modules can eliminate the requirement for seperate channel banks  or access gateways. Digiums commercial, toll-free quality High  Performance Echo Cancellation (HPEC) software is available to our analog  customers at no additional cost. The optional VPMADT032 hardware echo  cancellation module provides the same toll-quality HPEC but without the  perfoamce inpact of a software based solution. Scaling of an analog card  solution is accomplished by adding additional cards and FXO / FXS  modules.</p>
<p><strong>Core Features</strong></p>
<ul>
<li>4 Ports for connecting analog telephones or POTS lines</li>
<li>Half-length Analog Card</li>
<li>Up to 4 FXS or FXO Modules</li>
<li>High Performance Echo Cancellation (HPEC) Software (Optional)</li>
<li>TDM410 for use with a PCI 2.2 compliant slot</li>
</ul>
<p>888VoipStore.com: <a href="http://www.888voipstore.com/digium">Digium Products Page</a></p>
 ]]></content:encoded>
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		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Digium TDM800P Analog Interface Card Overview</title>
		<link>http://voipspeak.net/2010/03/digium-tdm800p-analog-interface-card-overview/</link>
		<comments>http://voipspeak.net/2010/03/digium-tdm800p-analog-interface-card-overview/#comments</comments>
		<pubDate>Tue, 16 Mar 2010 15:01:32 +0000</pubDate>
		<dc:creator>KerryG</dc:creator>
				<category><![CDATA[Feature]]></category>
		<category><![CDATA[Reviews]]></category>
		<category><![CDATA[Analog]]></category>
		<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Digium]]></category>
		<category><![CDATA[FXO]]></category>
		<category><![CDATA[FXS]]></category>

		<guid isPermaLink="false">http://voipspeak.net/?p=286</guid>
		<description><![CDATA[Kerry and the team at VoipStore.com have put together this great overview on the Digium TDM800P Analog cards. The Wildcard TDM800P is a half-length PCI 2.2-compliant, 8 port card for connecting analog telephones and analog POTS lines through a PC. It supports combinations of FXS and/or FXO modules for a total of 8 lines. The [...]]]></description>
			<content:encoded><![CDATA[<p><object id="wistia_88584" classid="clsid:d27cdb6e-ae6d-11cf-96b8-444553540000" width="560" height="315" codebase="http://download.macromedia.com/pub/shockwave/cabs/flash/swflash.cab#version=6,0,40,0"><param name="allowfullscreen" value="true" /><param name="allowscriptaccess" value="always" /><param name="wmode" value="opaque" /><param name="flashvars" value="playButtonVisible=true&amp;unbufferedSeek=true&amp;controlsVisibleOnLoad=false&amp;autoPlay=false&amp;videoUrl=http://embed.wistia.com/deliveries/b6678e3a4f8e5e2f594558bee38d9fa96fe8f66b.bin&amp;stillUrl=http://embed.wistia.com/deliveries/865e288c301212ca26d0f5526c0254040fea589e.bin&amp;embedServiceURL=http://distillery.wistia.com/x&amp;accountKey=wistia-production_542&amp;mediaID=wistia-production_88584&amp;mediaDuration=348" /><param name="src" value="http://embed.wistia.com/flash/embed_player_v1.1.swf" /><param name="name" value="wistia_88584" /><embed id="wistia_88584" type="application/x-shockwave-flash" width="560" height="315" src="http://embed.wistia.com/flash/embed_player_v1.1.swf" name="wistia_88584" flashvars="playButtonVisible=true&amp;unbufferedSeek=true&amp;controlsVisibleOnLoad=false&amp;autoPlay=false&amp;videoUrl=http://embed.wistia.com/deliveries/b6678e3a4f8e5e2f594558bee38d9fa96fe8f66b.bin&amp;stillUrl=http://embed.wistia.com/deliveries/865e288c301212ca26d0f5526c0254040fea589e.bin&amp;embedServiceURL=http://distillery.wistia.com/x&amp;accountKey=wistia-production_542&amp;mediaID=wistia-production_88584&amp;mediaDuration=348" wmode="opaque" allowscriptaccess="always" allowfullscreen="true"></embed></object></p>
<p>Kerry and the team at VoipStore.com have put together this great overview on the Digium TDM800P Analog cards.</p>
<p>The Wildcard TDM800P is a half-length PCI 2.2-compliant, 8 port card   for connecting analog telephones and analog POTS lines through a PC. It   supports combinations of FXS and/or FXO modules for a total of 8  lines.</p>
<p>The TDM800P provides an industry first with 8 standard two-wire,   RJ-11 interfaces on a single PCI bracket. This eliminates the need for   multiple brackets, external dongles, or splitters. In doing so, the   TDM800P reduces part complexity, cable clutter, and points of failure.<span id="more-286"></span></p>
<p>The  TDM800P contains two module banks. Each bank supports up to four  analog  interfaces. The module banks may be filled either with one  standard  Digium® quad analog module (<a href="http://www.888voipstore.com/digium-s400m-pr-16269.html">FXS &#8211;  S400M</a>, <a href="http://www.888voipstore.com/digium-x400m-pr-17004.html">FXO &#8211;  X400M</a>), or up to two standard Digium® single analog modules (<a href="http://www.888voipstore.com/digium-s110m-pr-16271.html">FXS- S110M</a>,  <a href="http://www.888voipstore.com/digium-x100m-pr-18863.html">FXO &#8211;  X100M</a>) enabling the creation of any combination of ports</p>
<p>The  optional hardware echo cancellation module provides 1024 taps  (128  milliseconds) of echo cancellation for superior echo cancellation  on  both FXO and FXS interfaces.</p>
<p>If the hardware echo cancellation module is not installed, the  TDM800P may be used in conjunction with Digium&#8217;s High Performance Echo  Cancellation (HPEC) software, a commercial and toll quality hybrid echo  cancellation  solution. Digium&#8217;s HPEC provides 16ms to 128ms of  selectable near-end  ITU G.168 compliant echo cancellation in software.</p>
<p>Using  this card in concert with Digium&#8217;s Asterisk® software,  standard PC  hardware, and the Linux® OS, you can create SME or SOHO  telephony  environments capable of satisfying the needs of small or  medium  business applications at an industry-leading price.</p>
<p>Digium Analog cards, with their interchangeable single and quad FXS  and FXO modules can eliminate the requirement for separate channel banks  or access gateways. Digium&#8217;s commercial, toll-free quality High  Performance Echo Cancellation (HPEC) software is available to our analog  customers at no additional cost. The optional VPMADT032 hardware echo  cancellation module provides the same toll-quality HPEC but without the performance impact of a software based solution. Scaling of an analog card  solution is accomplished by adding additional cards and FXO / FXS  modules.</p>
<ul>
<li>8 RJ-11 interfaces on a single PCI bracket</li>
<li>8 Ports for connecting analog telephones or POTS lines</li>
<li>Half-length Analog Card</li>
<li>Up to 4 Single FXS or FXO Modules, or 2 Quad FXS or FXO Modules</li>
<li>High Performance Echo Cancellation (HPEC) Software (Optional)</li>
<li>VoiceBus™ technology</li>
</ul>
<p>888VoipStore.com: <a href="http://www.888voipstore.com/digium">Digium Products Page</a></p>
 ]]></content:encoded>
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		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Digium unleashes new SwitchVox Appliances</title>
		<link>http://voipspeak.net/2009/10/digium-unleashes-new-switchvox-appliances/</link>
		<comments>http://voipspeak.net/2009/10/digium-unleashes-new-switchvox-appliances/#comments</comments>
		<pubDate>Thu, 01 Oct 2009 16:48:01 +0000</pubDate>
		<dc:creator>admin</dc:creator>
				<category><![CDATA[Feature]]></category>
		<category><![CDATA[News]]></category>
		<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Digium]]></category>
		<category><![CDATA[SwitchVox]]></category>
		<category><![CDATA[VoIP News]]></category>
		<category><![CDATA[VoIP Software]]></category>
		<category><![CDATA[VoIP Systems]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=196</guid>
		<description><![CDATA[New and improved versions of Digium’s award-winning Switchvox IP PBX appliances are now available.  The AA65, AA305 and AA355 models each feature a new front-mounted LCD control panel, making it easy to setup and manage your system.  Also, the AA65 features a new rack-friendly chassis design and an internal power supply – replacing the AA60 [...]]]></description>
			<content:encoded><![CDATA[<p><a href="http://www.voipspeak.net/wp-content/uploads/2009/10/switchvox365.jpg" rel="thumbnail"><img class="alignright size-medium wp-image-197" title="switchvox365" src="http://www.voipspeak.net/wp-content/uploads/2009/10/switchvox365-300x143.jpg" alt="switchvox365" width="300" height="143" /></a>New and improved versions of Digium’s award-winning Switchvox IP PBX appliances are now available.  The AA65, AA305 and AA355 models each feature a new front-mounted LCD control panel, making it easy to setup and manage your system.  Also, the AA65 features a new rack-friendly chassis design and an internal power supply – replacing the AA60 and its external power supply.  Each of these models are easily managed with the Switchvox web-based admin suite and Switchboard, Digium’s unique interface that makes it easy to see and manage your entire call network from a single screen.<span id="more-196"></span></p>
<p><strong><a href="http://www.voipspeak.net/wp-content/uploads/2009/10/switchvox205.jpg" rel="thumbnail"><img class="alignright size-medium wp-image-199" title="switchvox205" src="http://www.voipspeak.net/wp-content/uploads/2009/10/switchvox205-300x135.jpg" alt="switchvox205" width="300" height="135" /></a>Product Information:</strong></p>
<p>Switchvox SOHO remains offered only on the older model AA60 chassis.</p>
<p>Switchvox SMB on the AA60 has been fully replaced by Switchvox SMB on the AA65.</p>
<p>The AA300 has been fully replaced by the AA305.</p>
<p>The AA350 has been fully replaced by the AA355.</p>
<p><a href="http://www.voipspeak.net/wp-content/uploads/2009/10/switchvox65.jpg" rel="thumbnail"><img class="alignright size-medium wp-image-198" title="switchvox65" src="http://www.voipspeak.net/wp-content/uploads/2009/10/switchvox65-300x112.jpg" alt="switchvox65" width="300" height="112" /></a>The LCD panel offers the following options and menu organization:</p>
<ul>
<li>View System Info
<ul>
<li>Administer your PBX at https://IP/admin</li>
</ul>
</li>
<li>Configure Network
<ul>
<li>Active Interface
<ul>
<li>eth1</li>
<li>eth0</li>
</ul>
</li>
<li>Set DHCP / Static
<ul>
<li>DHCP</li>
<li>Static</li>
</ul>
</li>
<li>Change IP Address
<ul>
<li>192.168.0.20 (Note this is an example, not a default IP)</li>
</ul>
</li>
<li>Change Subnet mask
<ul>
<li>255.255.255.0 (Note this is an example, not a default subnet mask)</li>
</ul>
</li>
<li>Change Gateway
<ul>
<li>192.168.0.1 (Note this is an example, not a default gateway)</li>
</ul>
</li>
<li>Change DNS Server
<ul>
<li>192.168.0.100 (Note this is an example, not a default DNS server)</li>
</ul>
</li>
</ul>
</li>
<li>Reboot PBX
<ul>
<li>Yes / No</li>
</ul>
</li>
<li>Shutdown PBX
<ul>
<li>Yes / No</li>
</ul>
</li>
<li>Reset HTTPs Certificate
<ul>
<li>Yes / No</li>
</ul>
</li>
<li>Tech Support Access
<ul>
<li>Enable Yes / No</li>
</ul>
</li>
<li>Restore Web Access
<ul>
<li>Yes / No</li>
</ul>
</li>
<li>Reset Administrator Password</li>
<li>Change LCD Password</li>
</ul>
 ]]></content:encoded>
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		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Interview with Mike Storella from snom</title>
		<link>http://voipspeak.net/2009/07/interview-with-mike-storella-from-snom/</link>
		<comments>http://voipspeak.net/2009/07/interview-with-mike-storella-from-snom/#comments</comments>
		<pubDate>Fri, 10 Jul 2009 16:11:06 +0000</pubDate>
		<dc:creator>admin</dc:creator>
				<category><![CDATA[Feature]]></category>
		<category><![CDATA[News]]></category>
		<category><![CDATA[3CX]]></category>
		<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Digium]]></category>
		<category><![CDATA[snom]]></category>
		<category><![CDATA[trixbox]]></category>
		<category><![CDATA[VoIP News]]></category>
		<category><![CDATA[VoIP Software]]></category>
		<category><![CDATA[VoIP Systems]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=157</guid>
		<description><![CDATA[VoipStore.com, the content site that is run by 888VoipStore has a new article posted that is an interview with Mike Storella from snom. Mike talks about their products and where snom products are heading in the future. Interview with Mike Storella from snom Source: VoipStore.com]]></description>
			<content:encoded><![CDATA[<p><a href="http://voipstore.com/"></a><a href="http://www.voipspeak.net/wp-content/uploads/2009/07/snom.png" rel="thumbnail"><img class="alignright size-full wp-image-160" title="snom" src="http://www.voipspeak.net/wp-content/uploads/2009/07/snom.png" alt="snom" width="200" height="200" /></a>VoipStore.com, the content site that is run by <a href="http://888VoipStore.com">888VoipStore</a> has a new article posted that is an interview with Mike Storella from snom. Mike talks about their products and where snom products are heading in the future.</p>
<p><a title="Permanent Link to Interview with Mike Storella from snom" rel="bookmark" href="http://www.voipstore.com/2009/07/interview-with-mike-ostrander-from-snom/">Interview with Mike Storella from snom</a></p>
<p>Source: <a href="http://voipstore.com">VoipStore.com</a></p>
 ]]></content:encoded>
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		</item>
		<item>
		<title>Skype for Asterisk Announced</title>
		<link>http://voipspeak.net/2008/09/skype-for-asterisk-announced/</link>
		<comments>http://voipspeak.net/2008/09/skype-for-asterisk-announced/#comments</comments>
		<pubDate>Thu, 25 Sep 2008 20:32:32 +0000</pubDate>
		<dc:creator>admin</dc:creator>
				<category><![CDATA[News]]></category>
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		<category><![CDATA[VoIP Software]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=130</guid>
		<description><![CDATA[Waiting until the last day, Digium and Skype have announced a new channel driver that will soon go into beta testing. The new Skype for Asterisk will allow you to create a native Skype trunk within Asterisk 1.4 or 1.6. This new module will not be open source as it will require a per-channel license [...]]]></description>
			<content:encoded><![CDATA[<p>Waiting until the last day, Digium and Skype have announced a new channel driver that will soon go into beta testing. The new Skype for Asterisk will allow you to create a native Skype trunk within Asterisk 1.4 or 1.6. This new module will not be open source as it will require a per-channel license fee (pricing not yet set). Beta testing for a select group of users will begin shortly. The Skype module will be available through the Digium Asterisk Marketplace. The Skype channel will allow you to not only use Skype-In and Skype-Out minutes but will also allow you to make and receive free calls from Skype clients and other Skype channel users. Skype&#8217;s wideband codecs, that provide for excellent call quality that has helped make Skype successful will be built into the module to allow for transcoding between Asterisk codecs and the Skype SYLK codec. You will also be able to setup multiple Skype names on the channel to provide for routing within the Asterisk PBX based on the Skype handle that was called.</p>
<p>For more information, please visit <a href="http://astricon.net/skype" target="_blank">http://astricon.net/skype</a></p>
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		<title>OutCALL 1.5 Released (Outlook Integration)</title>
		<link>http://voipspeak.net/2008/06/outcall-15-released-outlook-integration/</link>
		<comments>http://voipspeak.net/2008/06/outcall-15-released-outlook-integration/#comments</comments>
		<pubDate>Fri, 13 Jun 2008 16:03:03 +0000</pubDate>
		<dc:creator>admin</dc:creator>
				<category><![CDATA[News]]></category>
		<category><![CDATA[Digium]]></category>
		<category><![CDATA[SwitchVox]]></category>
		<category><![CDATA[VoIP Hardware]]></category>
		<category><![CDATA[VoIP News]]></category>
		<category><![CDATA[VoIP Software]]></category>
		<category><![CDATA[VoIP Systems]]></category>

		<guid isPermaLink="false">http://www.voipspeak.net/?p=8</guid>
		<description><![CDATA[OutCALL was designed as a commercial appplication allowing Asterisk users integration with Microsoft Outlook with placing and receiving phone calls. Changes for version 1.5 include: Added support for UNICODE OutCALL now uses Extended MAPI to load contacts from Outlook Added Outlook Import Rules Added support for calls coming from Queue Source: OutCALL]]></description>
			<content:encoded><![CDATA[<p><a href="http://www.voipspeak.net/wp-content/uploads/2008/06/outcallscreen.png" rel="thumbnail"><img class="alignnone size-thumbnail wp-image-9 alignright" style="float: right;" title="outcallscreen" src="http://www.voipspeak.net/wp-content/uploads/2008/06/outcallscreen-150x150.png" alt="" width="150" height="150" /></a>OutCALL was designed as a commercial appplication allowing Asterisk users integration with Microsoft Outlook with placing and receiving phone calls. Changes for version 1.5 include:</p>
<ul>
<li>Added support for UNICODE</li>
<li>OutCALL now uses Extended MAPI to load contacts from Outlook</li>
<li>Added Outlook Import Rules</li>
<li>Added support for calls coming from Queue</li>
</ul>
<p>Source: <a href="http://outcall.sourceforge.net/" target="_blank">OutCALL</a></p>
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		<title>Digium aquires SwitchVox</title>
		<link>http://voipspeak.net/2007/09/digium-aquires-switchvox/</link>
		<comments>http://voipspeak.net/2007/09/digium-aquires-switchvox/#comments</comments>
		<pubDate>Fri, 28 Sep 2007 03:55:27 +0000</pubDate>
		<dc:creator>admin</dc:creator>
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		<guid isPermaLink="false">http://www.voipspeak.net/?p=35</guid>
		<description><![CDATA[PHOENIX, Ariz. (AstriCon 2007) — September 27, 2007 — Digium®, Inc., the Asterisk® Company, today announced it has acquired Switchvox, a leading provider of IP PBX phone systems for small- and medium-sized businesses (SMBs). The acquisition bolsters Digium&#8217;s presence in the SMB market and provides a strong platform on which to advance its Asterisk-based unified [...]]]></description>
			<content:encoded><![CDATA[<p>PHOENIX, Ariz. (AstriCon 2007) — September 27, 2007 — Digium®, Inc., the Asterisk® Company, today announced it has acquired Switchvox, a leading provider of IP PBX phone systems for small- and medium-sized businesses (SMBs). The acquisition bolsters Digium&#8217;s presence in the SMB market and provides a strong platform on which to advance its Asterisk-based unified communications solution.</p>
<p>Digium is the creator and driving force behind Asterisk, the world&#8217;s most popular and successful open source communications platform with an estimated 3.5 million servers worldwide providing VoIP calls for businesses and residential end users. More resellers, telecom professionals and software developers choose Digium&#8217;s products than those of any other open source telephony company because only Digium delivers the technical superiority, security and flexibility associated with genuine Asterisk.</p>
<p>Switchvox, with an estimated 65,000 end points in operation, is the world&#8217;s largest and most successful supplier of open source-based IP PBX products for businesses. The combination of Digium and Switchvox will provide open source-based products and solutions that are unrivaled in the industry.</p>
<p>&#8220;Switchvox has built a successful business by offering simple solutions that make communication easier while Digium is known the world over as the creator and primary maintainer of the Asterisk open source telephony code,&#8221; said Danny Windham, president and CEO of Digium. &#8220;This acquisition is an ideal pairing of two companies committed to providing Asterisk-based telephony solutions that are less expensive, easier to use and more flexible than other telephony products on the market.&#8221;</p>
<p>Digium and Switchvox will begin integration immediately and plan to unveil a new product strategy and roadmap later this year focused on giving SMBs and enterprises new choices in Asterisk-based unified communications solutions.</p>
<p>&#8220;The entire Switchvox team is excited to become part of the Digium family as it continues to gain momentum and win market share from traditional phone system vendors,&#8221; said Joshua Stephens, CEO of Switchvox. &#8220;We look forward to working together to create unified communications offerings that redefine the VoIP industry and give SMB and enterprise customers advanced solutions at price points that are undeniable.&#8221;</p>
<p>About Digium<br />
Digium®, Inc., the Asterisk® Company, created, owns and is the innovative force behind Asterisk, the most widely used open source telephony software. Since its founding in 1999, Digium has grown to become the open source alternative to the traditional communication providers, with offerings that cost as much as 80 percent less than proprietary voice communication platforms. Digium makes Asterisk available to the open source community under the GNU General Public License (GPL) and uses business-class Asterisk to power a broad family of products for small, medium and large businesses. The company&#8217;s product line also includes a wide range of hardware to enable resellers and customers to design their own voice over IP (VoIP) systems. This year alone, Asterisk will support over 12 billion minutes of calls. Additional information is available at <a href="http://www.digium.com/">www.digium.com</a>.</p>
<p>The Digium logo, Digium, Asterisk, Asterisk Business Edition, AsteriskNOW, Asterisk Appliance and the Asterisk logo are trademarks of Digium, Inc. All other trademarks are property of their respective owners.</p>
<p># # #</p>
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