<?xml version="1.0" encoding="UTF-8"?>
<?xml-stylesheet type="text/xsl" media="screen" href="/~d/styles/rss2full.xsl"?><?xml-stylesheet type="text/css" media="screen" href="http://feeds.feedburner.com/~d/styles/itemcontent.css"?><rss xmlns:content="http://purl.org/rss/1.0/modules/content/" xmlns:wfw="http://wellformedweb.org/CommentAPI/" xmlns:dc="http://purl.org/dc/elements/1.1/" xmlns:atom="http://www.w3.org/2005/Atom" xmlns:geo="http://www.w3.org/2003/01/geo/wgs84_pos#" xmlns:feedburner="http://rssnamespace.org/feedburner/ext/1.0" version="2.0">

<channel>
	<title>Inside the Asterisk</title>
	
	<link>http://blogs.digium.com</link>
	<description>A little slice o' Digium.</description>
	<pubDate>Mon, 29 Jun 2009 12:56:48 +0000</pubDate>
	<generator>http://wordpress.org/?v=2.6.1</generator>
	<language>en</language>
			<geo:lat>34.748066</geo:lat><geo:long>-86.683175</geo:long><atom10:link xmlns:atom10="http://www.w3.org/2005/Atom" rel="self" href="http://feeds.feedburner.com/InsideTheAsterisk" type="application/rss+xml" /><atom10:link xmlns:atom10="http://www.w3.org/2005/Atom" rel="hub" href="http://pubsubhubbub.appspot.com" /><item>
		<title>Had me a Blast</title>
		<link>http://feedproxy.google.com/~r/InsideTheAsterisk/~3/afOBzGilUIk/</link>
		<comments>http://blogs.digium.com/2009/06/29/had-me-a-blast/#comments</comments>
		<pubDate>Mon, 29 Jun 2009 12:56:48 +0000</pubDate>
		<dc:creator>Malcolm Davenport</dc:creator>
		
		<category><![CDATA[Digium]]></category>

		<guid isPermaLink="false">http://blogs.digium.com/?p=4373</guid>
		<description><![CDATA[Summer Digium, had me a blast,
Summer Digium, happened so fast.
Those su-hum-mer, nigh-heigh-eigh-heights!
Summer at Digium means one thing, heat.  We&#8217;ve been experiencing a heat wave lately and it&#8217;s been so hot outside that the outdoor picnic table remains unused:

And that everyone understands the merits of remaining cool with shorts, some shoes, and lots of liquids while [...]]]></description>
			<content:encoded><![CDATA[<p>Summer Digium, had me a blast,</p>
<p>Summer Digium, happened so fast.</p>
<p>Those su-hum-mer, nigh-heigh-eigh-heights!</p>
<a href="http://blogs.digium.com/2009/06/29/had-me-a-blast/"><em>Click here to view the embedded video.</em></a>
<p>Summer at Digium means one thing, heat.  We&#8217;ve been experiencing a heat wave lately and it&#8217;s been so hot outside that the outdoor picnic table remains unused:</p>
<p><a href="http://blogs.digium.com/wp-content/uploads/2009/06/photo2.jpg"><img class="alignnone size-medium wp-image-4533" title="picnic table" src="http://blogs.digium.com/wp-content/uploads/2009/06/photo2-300x225.jpg" border="0" alt="" width="300" height="225" /></a></p>
<p>And that everyone understands the merits of remaining cool with shorts, some shoes, and lots of liquids while indoors:</p>
<p><a href="http://blogs.digium.com/wp-content/uploads/2009/06/photo3.jpg"><img class="alignnone size-medium wp-image-4543" title="shorts" src="http://blogs.digium.com/wp-content/uploads/2009/06/photo3-300x225.jpg" border="0" alt="" width="300" height="225" /></a></p>
<p>Meanwhile, with the madness of some of our product launches this year already out of the way, it&#8217;s business as usual (working towards future product releases) over in my unkempt office:</p>
<p><a href="http://blogs.digium.com/wp-content/uploads/2009/06/photo.jpg"><img class="alignnone size-medium wp-image-4523" title="malcolm office" src="http://blogs.digium.com/wp-content/uploads/2009/06/photo-300x225.jpg" border="0" alt="" width="300" height="225" /></a></p>
<p>And, the Digium Softball team posted its best half-season record yet; 5-2 in division E of the <a title="RPMSL" href="http://www.eteamz.com/rpmsl-hsv/standings/index.cfm?season=321411&amp;cat=525096&amp;xdiv=1&amp;division=3140618&amp;show=schedule&amp;tteam=5084558&amp;sort=12&amp;sortorder=0" target="_blank">Research Park Mixed Softball league</a>.  Congratulations, team!</p>
<p>What&#8217;s going on in your office this summer?</p>
<img src="http://feeds.feedburner.com/~r/InsideTheAsterisk/~4/afOBzGilUIk" height="1" width="1"/>]]></content:encoded>
			<wfw:commentRss>http://blogs.digium.com/2009/06/29/had-me-a-blast/feed/</wfw:commentRss>
		<feedburner:origLink>http://blogs.digium.com/2009/06/29/had-me-a-blast/</feedburner:origLink></item>
		<item>
		<title>Asterisk: Always On</title>
		<link>http://feedproxy.google.com/~r/InsideTheAsterisk/~3/IcFSr4v44dE/</link>
		<comments>http://blogs.digium.com/2009/06/23/asterisk-always-on/#comments</comments>
		<pubDate>Tue, 23 Jun 2009 13:32:47 +0000</pubDate>
		<dc:creator>jtodd</dc:creator>
		
		<category><![CDATA[Digium]]></category>

		<category><![CDATA[Open Source]]></category>

		<category><![CDATA[mobile]]></category>

		<category><![CDATA[android]]></category>

		<category><![CDATA[Asterisk]]></category>

		<category><![CDATA[embedded software]]></category>

		<category><![CDATA[handheld]]></category>

		<category><![CDATA[iphone]]></category>

		<category><![CDATA[mobile phone]]></category>

		<category><![CDATA[radio]]></category>

		<category><![CDATA[real-time communications]]></category>

		<guid isPermaLink="false">http://blogs.digium.com/?p=4413</guid>
		<description><![CDATA[
In the last 15 years, there has been a significant shift in the focus of development from the concept of &#8220;compute&#8221; to &#8220;interact&#8221;.  While there is certainly a spectacular improvement in the ability of machines to store, calculate, and interpret data, I would suggest that the ability of individuals to interact with and easily access, [...]]]></description>
			<content:encoded><![CDATA[<div>
<p>In the last 15 years, there has been a significant shift in the focus of development from the concept of &#8220;compute&#8221; to &#8220;interact&#8221;.  While there is certainly a spectacular improvement in the ability of machines to store, calculate, and interpret data, I would suggest that the ability of individuals to interact with and easily access, understand, and modify that data is that which has been the most powerful factor in the spread of information systems into all facets of life.</p>
<p>There are four basic methods that people use to interact with computing systems: eyes (seeing screens), hands (keyboards, mice, or other I/O systems), ears, and mouth (spoken commands.)  There is currently a wealth of methods for interaction with eyes and hands - humans are primarily visual animals, so this comes as no surprise.  And humans are marvelous communicators verbally - we convey most of our sophisticated information in spoken language (even these words you see on the screen, a visual interface, are artifacts of our ability to talk.)  The first widespread technology that many people were exposed to, and certainly the largest network that has been built, is the telephone system.  But there still remains a gap between the screen-based world and the sound-based world.  Using spoken language is our first choice for communications, but unless the other person is standing within earshot it still takes a back seat to visual communications methods when computers are involved.  I think this is about to change for the better.  Voice interaction will not replace visual, but Asterisk is enabling it to close the gap a bit by providing a useful voice toolkit to cutting-edge developers as well as back-office pragmatists.  It&#8217;s easy and fast to get Asterisk bolted on to your existing application.  The only thing that is trailing a bit are control methods for devices (mobile phones) to allow quick access to microphone/speakers, and similar control planes for desktop machines that would allow transparent direction of audio in a secure way.  All of the basic components exist, so now we&#8217;re down to politics of device control.  Having &#8220;always on&#8221; communications via Asterisk that is available without even touching a keypad or looking at a screen - that&#8217;s what I envision happening, given what I can see from some of the devices and directions that are evident.</p>
<p>Earlier this year as part of a conference, I had the chance to use some of the &#8220;secret-service&#8221; type radio connections, with a discreet headset and a lapel microphone.  This was an eye-opening experience - it was amazingly productive!  I&#8217;m very familiar with PTT phones and I&#8217;ve never been very impressed, but a ubiquitous device that is basically &#8220;implanted&#8221; in the ear and had selectable broadcast/unicast capabilities was surprisingly useful in a small working group.  No fumbling for the device, fiddling with the screen or keypad, no &#8220;broadcast&#8221; messages that everyone in the room could hear.  The combination of this type of system bolted into Asterisk with voice-recognition systems and scripts - THAT would be great.  And easy.  And inexpensive.  There is a whole industry growing around those kinds of tools, and Asterisk is the ideal platform to bridge between existing screen-based tools and their audio counterparts.  (Yes, I know about some of the tools that exist already, but they&#8217;re typically in the $XXXk range for systems like Vocera - when they&#8217;re $0k for Asterisk, then innovation will really heat up.)</p>
<p>Already Asterisk is being grafted onto real-time communications tools.  Google search found <a title="Asterisk Radio Network" href="http://asteriskradio.net/wp/" target="_blank">Asterisk Radio Networks</a> and <a title="WeComm" href="http://kb9mwr.blogspot.com/2009/05/wecomm-digital-audio-repeater-linking.html" target="_blank">Wisconsin Emergency Communications (WeComm</a>) on my first keyword entry.  This is GREAT stuff, and I&#8217;m sure there are many more small projects out there linking new hardware and new software together with Asterisk as the glue.  Radios are great, but they&#8217;re a niche - the real target is the mobile device.</p>
<p>There are iPhone and Android development efforts to graft Asterisk connection methods into mobile devices.  Android has great promise for replacing the peer-to-peer radio model with a network model.  There are rumors (you know who you are) about highly-integrated free Flash plugins for Asterisk being developed by third-party OSS developers.  Getting into the code under the hood on Android and letting it work more seamlessly with VoIP systems built on Asterisk - that will be fun to watch.  And it&#8217;s happening already.  Hang on, if you think the volume is loud now about Asterisk being built into everything, the next year is going to be deafening!</p>
<p>JT</p></div>
<img src="http://feeds.feedburner.com/~r/InsideTheAsterisk/~4/IcFSr4v44dE" height="1" width="1"/>]]></content:encoded>
			<wfw:commentRss>http://blogs.digium.com/2009/06/23/asterisk-always-on/feed/</wfw:commentRss>
		<feedburner:origLink>http://blogs.digium.com/2009/06/23/asterisk-always-on/</feedburner:origLink></item>
		<item>
		<title>Digium calls for 3rd Annual Asterisk Innovation Award Entries</title>
		<link>http://feedproxy.google.com/~r/InsideTheAsterisk/~3/zamjmuggBL0/</link>
		<comments>http://blogs.digium.com/2009/06/14/digium-calls-for-3rd-annual-asterisk-innovation-award-entries/#comments</comments>
		<pubDate>Sun, 14 Jun 2009 23:07:53 +0000</pubDate>
		<dc:creator>beelinebill</dc:creator>
		
		<category><![CDATA[Asterisk Awards]]></category>

		<category><![CDATA[AstriCon]]></category>

		<category><![CDATA[Channel]]></category>

		<category><![CDATA[Community]]></category>

		<category><![CDATA[Digium]]></category>

		<guid isPermaLink="false">http://blogs.digium.com/?p=3793</guid>
		<description><![CDATA[Digium is pleased to be announce that we are accepting entries for our 3rd annual Asterisk Innovation Award program. This program was created to recognize Asterisk developers, customers and partners for outstanding achievements that are improving business processes, overcoming technology challenges and delivering record return on investment. Over the years, Mark Spencer, creator and author [...]]]></description>
			<content:encoded><![CDATA[<div style="margin: 0in 0in 0pt;">Digium is pleased to be announce that we are accepting entries for our 3rd annual Asterisk Innovation Award program. This program was created to recognize Asterisk developers, customers and partners for outstanding achievements that are improving business processes, overcoming technology challenges and delivering record return on investment. Over the years, Mark Spencer, creator and author of Asterisk and Digium&#8217;s CTO, has enjoyed sharing innovation in the Asterisk Community with audiences around the world.  Some of these innovations were so much fun and rewarding, 3 years ago we built the Innovation Award program around these ideas to share the success with millions of readers. Asterisk has been referred to as a &#8220;Swiss Army Knife&#8221; of IP Telephony because in it&#8217;s source code form it can be applied to endless applications from basic PBX, advanced clustered phone system, Call Center, and voice mail systems to plant watering systems, call center &#8216;quiz&#8217; questions to move up or down while on hold in the call queue, smart parking in the city to locate vacant parking spaces and many more.</div>
<p></p>
<div style="margin: 0in 0in 0pt;">Awards will be recognized in four distinct categories, each showcasing a particular area of innovation. The four awards are as follows:</div>
<p><strong><br />Pioneer Award :</strong> Does your application break new ground where no one has ever gone before? Do you integrate applications in a unique way that exhibits innovative uses of Asterisk? The award in this category is given to the entry that displays these attributes.</p>
<p><strong>Big Biz Asterisk Award :</strong> This category is reserved for the largest Asterisk-based solutions in enterprise-class businesses. Over 14% of all Asterisk-based installations have over 500 users and over 9% have over 1000 users. Share your success with the world.</p>
<p><strong>ROI Award :</strong> Does your Asterisk-based system provide a strong measurable return on investment second to none? Each year we uncover incredible ROI examples only available using open source Asterisk. </p>
<p><strong>Inside Out Award :</strong> Asterisk has a whimsical side unlike any other telephony platform and each year we find that one unique award-worthy application. This category is for applications outside of the typical telecommunications world.</p>
<div style="margin: 0in 0in 0pt;">Winners from each category will be announced October 13-15 at AstriCon 2009, in Glendale, Arizona. Digium Innovation Award winners will receive a congratulatory press release from Digium, Inc., a listing on the Digium Web site, and recognition by the industry, community, family and friends as well as at Astricon 2009, <a href="http://www.astricon.net">AstriCon ‘09</a>.</div>
<p></p>
<div style="margin: 0in 0in 0pt;">2008 Winners of the Innovation Awards include Greenfield Tech for Pioneer category, Integrics, Ltd for Big Business category, Emdeon Business Services for ROI category, PhonePlay for &#8220;Inside Out&#8221; category, and NTT Software received an honorable mention. Read more about last years winners <a href="http://www.digium.com/en/company/awards/archive/2008">here</a>.</div>
<p></p>
<div style="margin: 0in 0in 0pt;">2007 winners of the Innovation Awards include <!--ZZZLinkBegZZZ-->Aheeva for the Big Business category, OneBizTone for the ROI/Cost Savings category, Shelton|Johns for the Pioneer category, iPLATEu for the Most Unique category and Super Technologies’ DIDX . Read more <a href="http://www.digium.com/en/company/awards/archive/2007">here</a>.</div>
<p>

<div style="margin: 0in 0in 0pt;">
<p>To submit your 2009 Innovation Award entry application click <a href="http://www.digium.com/en/company/awards/innovation.php">here.</a></div>
</p>
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p>We look forward to hearing from you!</p>
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" />
<p><!--Session data--></p>
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<input id="gwProxy" type="hidden" /><!--Session data--><br />
<input id="jsProxy" onclick="jsCall();" type="hidden" />
<img src="http://feeds.feedburner.com/~r/InsideTheAsterisk/~4/zamjmuggBL0" height="1" width="1"/>]]></content:encoded>
			<wfw:commentRss>http://blogs.digium.com/2009/06/14/digium-calls-for-3rd-annual-asterisk-innovation-award-entries/feed/</wfw:commentRss>
		<feedburner:origLink>http://blogs.digium.com/2009/06/14/digium-calls-for-3rd-annual-asterisk-innovation-award-entries/</feedburner:origLink></item>
		<item>
		<title>Video Walkthrough: Switchvox Developer Central</title>
		<link>http://feedproxy.google.com/~r/InsideTheAsterisk/~3/-Gfxu5JzGbs/</link>
		<comments>http://blogs.digium.com/2009/05/21/video-walkthrough-switchvox-developer-central/#comments</comments>
		<pubDate>Thu, 21 May 2009 16:53:27 +0000</pubDate>
		<dc:creator>tristan</dc:creator>
		
		<category><![CDATA[Digium]]></category>

		<guid isPermaLink="false">http://blogs.digium.com/?p=3763</guid>
		<description><![CDATA[This video is a quick walkthrough of the Test Suite inside the brand new Switchvox Developer Central. It shows how to use this online tool to connect to your Switchvox and request XML via the new Extend API in Switchvox SMB 4.0.
]]></description>
			<content:encoded><![CDATA[<p>This video is a quick walkthrough of the <a title="Test Suite" href="http://developers.digium.com/switchvox/?pageView=testSuite">Test Suite</a> inside the brand new <a title="Switchvox Developer Central" href="http://developers.digium.com/switchvox">Switchvox Developer Central</a>. It shows how to use this online tool to connect to your Switchvox and request XML via the new Extend API in Switchvox SMB 4.0.</p>
<a href="http://blogs.digium.com/2009/05/21/video-walkthrough-switchvox-developer-central/"><em>Click here to view the embedded video.</em></a>
<img src="http://feeds.feedburner.com/~r/InsideTheAsterisk/~4/-Gfxu5JzGbs" height="1" width="1"/>]]></content:encoded>
			<wfw:commentRss>http://blogs.digium.com/2009/05/21/video-walkthrough-switchvox-developer-central/feed/</wfw:commentRss>
		<feedburner:origLink>http://blogs.digium.com/2009/05/21/video-walkthrough-switchvox-developer-central/</feedburner:origLink></item>
		<item>
		<title>Congrats to Google SoC students working on Asterisk</title>
		<link>http://feedproxy.google.com/~r/InsideTheAsterisk/~3/lhjWKTc911U/</link>
		<comments>http://blogs.digium.com/2009/05/04/congrats-to-google-soc-students-working-on-asterisk/#comments</comments>
		<pubDate>Mon, 04 May 2009 18:05:19 +0000</pubDate>
		<dc:creator>roderickm</dc:creator>
		
		<category><![CDATA[Asterisk Global Online Community]]></category>

		<category><![CDATA[Community]]></category>

		<category><![CDATA[Open Source]]></category>

		<guid isPermaLink="false">http://blogs.digium.com/?p=3713</guid>
		<description><![CDATA[Each summer since 2005, the Google Summer of Code offers stipends to student developers so they may contribute to open source software projects. The Asterisk project is again one of 150 accepted mentoring organizations. We offer hearty congratulations to the four students accepted into GSoC to contribute to Asterisk!
Brett Bryant and Eliel Sardañons will be [...]]]></description>
			<content:encoded><![CDATA[<p>Each summer since 2005, the Google Summer of Code offers stipends to student developers so they may contribute to open source software projects. The Asterisk project is again one of 150 accepted mentoring organizations. We offer hearty congratulations to the <a href="http://socghop.appspot.com/org/home/google/gsoc2009/asterisk">four students accepted</a> into GSoC to contribute to Asterisk!</p>
<p><strong>Brett Bryant</strong> and <strong>Eliel Sardañons</strong> will be building a common infrastructure to read and write configuration and other run-time data in Asterisk modules.</p>
<p><strong>Claude Patry</strong> will be adding CLI filtering features to assist with live debugging.</p>
<p><strong>Giuseppe Sucameli</strong> will be extending voicemail capabilities to make menus and voicemail behavior customizable.</p>
<p>Open Source Team Lead Russell Bryant <a href="http://lists.digium.com/pipermail/asterisk-dev/2009-April/038028.html">posted more detailed project descriptions</a> on the Asterisk-Dev mailing list.</p>
<p>According to the <a href="http://socghop.appspot.com/document/show/program/google/gsoc2009/faqs#goals">GSoC 2009 FAQ</a>, <em>Google Summer of Code</em> has several goals:</p>
<ul>
<li>Get more open source code created and released for the benefit of all</li>
<li>Inspire young developers to begin participating in open source development</li>
<li>Help open source projects identify and bring in new developers and committers</li>
<li>Provide students the opportunity to do work related to their academic pursuits during the summer (think &#8220;flip bits, not burgers&#8221;)</li>
<li>Give students more exposure to real-world software development scenarios (e.g., distributed development, software licensing questions, mailing-list etiquette)</li>
</ul>
<p>The program represents Google&#8217;s growing commitment of funding and administration that has a significant impact on the open source community. In addition to new code contributions, it engages hundreds of talented students with thriving open source development. Thanks to the mentors committed to guiding these students, and to Google for the Summer of Code program.</p>
<img src="http://feeds.feedburner.com/~r/InsideTheAsterisk/~4/lhjWKTc911U" height="1" width="1"/>]]></content:encoded>
			<wfw:commentRss>http://blogs.digium.com/2009/05/04/congrats-to-google-soc-students-working-on-asterisk/feed/</wfw:commentRss>
		<feedburner:origLink>http://blogs.digium.com/2009/05/04/congrats-to-google-soc-students-working-on-asterisk/</feedburner:origLink></item>
		<item>
		<title>Asterisk says “Hello” to Fax</title>
		<link>http://feedproxy.google.com/~r/InsideTheAsterisk/~3/f3akUuNN9Tk/</link>
		<comments>http://blogs.digium.com/2009/04/06/asterisk-says-hello-to-fax/#comments</comments>
		<pubDate>Mon, 06 Apr 2009 11:08:32 +0000</pubDate>
		<dc:creator>Malcolm Davenport</dc:creator>
		
		<category><![CDATA[Digium]]></category>

		<guid isPermaLink="false">http://blogs.digium.com/?p=3213</guid>
		<description><![CDATA[If you ask Google about faxing for Asterisk, with the search keywords asterisk and fax, and you ask Google to omit similar entries, you&#8217;ll end up with 52 pages of results.
If you ask Google how many times fax has been mentioned on an Asterisk mailing list, by setting the site parameter to lists.digium.com, then Google [...]]]></description>
			<content:encoded><![CDATA[<p>If you ask <a href="http://www.google.com">Google</a> about faxing for Asterisk, with the search keywords asterisk and fax, and you ask Google to omit similar entries, you&#8217;ll end up with 52 pages of results.</p>
<p>If you ask Google how many times fax has been mentioned on an <a href="http://www.asterisk.org">Asterisk</a> mailing list, by setting the site parameter to <a href="http://lists.digium.com">lists.digium.com</a>, then Google tells you there are 1120 utterances.</p>
<p>Yesterday, if you asked <a href="http://www.digium.com">Digium</a> for help in faxing documents through Asterisk, we&#8217;d have apologized and said that we didn&#8217;t offer a fax solution for Asterisk.</p>
<p>That was yesterday.</p>
<p>Today, Digium is pleased to announce <a href="http://www.digium.com/en/products/software/faxforasterisk.php">Fax For Asterisk</a>.</p>
<p>Fax For Asterisk is a commercial facsimile (Fax) termination and origination solution designed to enhance the capabilities of Open Source and commercial Asterisk as well as <a href="http://www.digium.com/switchvox">Switchvox</a>. Fax For Asterisk bundles a suite of user-friendly Asterisk applications and a licensed version of the industry&#8217;s leading fax modem software from <a href="http://www.commetrex.com/">Commetrex</a>. Fax For Asterisk provides low speed (14400bps) PSTN faxing via DAHDI-compatible telephony boards as well as VoIP faxing to T.38-compatible SIP endpoints and service providers. Licensed on a per-channel basis, Digium&#8217;s Fax For Asterisk provides a complete, cost-effective, commercial fax solution for Asterisk users.</p>
<p>Wait, I&#8217;ve forgotten something&#8230;okay, not really.  There&#8217;s also Free Fax For Asterisk.  Free Fax For Asterisk provides a single-channel only, per Asterisk, version of Fax For Asterisk, for free.  Want to use Free Fax For Asterisk now?  Visit the <a href="https://store.digium.com/productview.php?product_code=804-00007">Digium webstore</a> and get a license, free of charge.</p>
<img src="http://feeds.feedburner.com/~r/InsideTheAsterisk/~4/f3akUuNN9Tk" height="1" width="1"/>]]></content:encoded>
			<wfw:commentRss>http://blogs.digium.com/2009/04/06/asterisk-says-hello-to-fax/feed/</wfw:commentRss>
		<feedburner:origLink>http://blogs.digium.com/2009/04/06/asterisk-says-hello-to-fax/</feedburner:origLink></item>
		<item>
		<title>Digium Launches Support For Open Source Asterisk</title>
		<link>http://feedproxy.google.com/~r/InsideTheAsterisk/~3/oJcTTwAwwFM/</link>
		<comments>http://blogs.digium.com/2009/03/31/digium-launches-support-for-open-source-asterisk/#comments</comments>
		<pubDate>Tue, 31 Mar 2009 17:16:25 +0000</pubDate>
		<dc:creator>ssokol</dc:creator>
		
		<category><![CDATA[Digium]]></category>

		<guid isPermaLink="false">http://blogs.digium.com/?p=3573</guid>
		<description><![CDATA[Way back in September of 2008 (right about the time that the global economy was starting to go over the falls) Digium announced a number of initiatives, including commercial support for open source Asterisk.  It took a bit of time and effort to get all of our ducks into rows, but as of today we&#8217;re [...]]]></description>
			<content:encoded><![CDATA[<p>Way back in September of 2008 (right about the time that the global economy was starting to go over the falls) Digium announced a number of initiatives, including commercial support for open source Asterisk.  It took a bit of time and effort to get all of our ducks into rows, but as of today we&#8217;re open for business.  Asterisk support subscriptions are now available.  Head over to the <a title="Asterisk Support" href="http://www.digium.com/en/supportcenter/asterisk.php" target="_self">product page</a> for the full details.</p>
<p>So why did Digium change a long standing policy (providing support only Asterisk Business Edition) and take on the rather daunting challenge of supporting open source Asterisk?  Because that&#8217;s what our customers want.  When Asterisk hit the 1.0 stage back in 2004, most businesses had no idea what to make of open source.  Linux had gained some traction in the enterprise market, but applications (including voice communications) were still largely proprietary and closed.  Most CIOs and CFOs weren&#8217;t ready to bet the company (and their jobs) on open source.</p>
<p>Oh what a difference a few years and a global financial meltdown can make.  Since the 1.0 launch in the fall of &#8216;04 the CIOs of the world have seen open source evolve from a philosophical movement into a solid, respected and trusted method of collaboratively building value.  Just as importantly, those CIOs have seen budgets shrink drastically.  In only five years (and three major releases of Asterisk) the pendulum has shifted towards open source and and away from bloated, vendor-locked proprietary systems.  Open is the new safe bet.</p>
<p>So all that brings us back to support for Asterisk.  The market talked, we listened, and today we&#8217;re ready to take calls, answer questions and do whatever needs to be done to make Asterisk work for our customers.  It is now safe to step into the Open.</p>
<img src="http://feeds.feedburner.com/~r/InsideTheAsterisk/~4/oJcTTwAwwFM" height="1" width="1"/>]]></content:encoded>
			<wfw:commentRss>http://blogs.digium.com/2009/03/31/digium-launches-support-for-open-source-asterisk/feed/</wfw:commentRss>
		<feedburner:origLink>http://blogs.digium.com/2009/03/31/digium-launches-support-for-open-source-asterisk/</feedburner:origLink></item>
		<item>
		<title>Seven Steps to Better SIP Security with Asterisk</title>
		<link>http://feedproxy.google.com/~r/InsideTheAsterisk/~3/PBVJodflz_Y/</link>
		<comments>http://blogs.digium.com/2009/03/28/sip-security/#comments</comments>
		<pubDate>Sun, 29 Mar 2009 00:25:23 +0000</pubDate>
		<dc:creator>jtodd</dc:creator>
		
		<category><![CDATA[Digium]]></category>

		<guid isPermaLink="false">http://blogs.digium.com/?p=3353</guid>
		<description><![CDATA[In case any of you were wondering why there has been a fairly notable upswing in the attacks happening on SIP endpoints, the answer is &#8220;script kiddies.&#8221;  In the last few months, a number of new tools have made it easy for knuckle-draggers to attack and defraud SIP endpoints, Asterisk-based systems included.  There are easily-available [...]]]></description>
			<content:encoded><![CDATA[<div>In case any of you were wondering why there has been a fairly notable upswing in the attacks happening on SIP endpoints, the answer is &#8220;script kiddies.&#8221;  In the last few months, a number of new tools have made it easy for knuckle-draggers to attack and defraud SIP endpoints, Asterisk-based systems included.  There are easily-available tools that scan networks looking for SIP hosts, and then scan hosts looking for valid extensions, and then scan valid extensions looking for passwords.  You can take steps, NOW, to eliminate many of these problems.  I think the community is interested in coming up with an integrated Asterisk-based solution that is much wider in scope for dynamic protection (community-shared blacklists is the current thinking) but that doesn&#8217;t mean you should wait for some new tool to defend your systems.  You can IMMEDIATELY take fairly common-sense measures to protect your Asterisk server from the bulk of the scans and attacks that are on the increase. The methods and tools for protection already exists - just apply them, and you&#8217;ll be able to sleep more soundly at night.</div>
<p> </p>
<div><strong>Seven Easy Steps to Better SIP Security on Asterisk:</strong></div>
<p> </p>
<div style="padding-left: 30px;"><strong>1) Don&#8217;t accept SIP authentication requests from all IP addresses.</strong>  Use the &#8220;permit=&#8221; and &#8220;deny=&#8221; lines in sip.conf to only allow a reasonable subset of IP addresess to reach each listed extension/user in your sip.conf file.  Even if you accept inbound calls from &#8220;anywhere&#8221; (via [default]) don&#8217;t let those users reach authenticated elements!</div>
<p> </p>
<div style="padding-left: 30px;"><strong>2) Set &#8220;alwaysauthreject=yes&#8221; in your sip.conf file.</strong>  This option has been around for a while (since 1.2?) but the default is &#8220;no&#8221;, which allows extension information leakage.  Setting this to &#8220;yes&#8221; will reject bad authentication requests on valid usernames with the same rejection information as with invalid usernames, denying remote attackers the ability to detect existing extensions with brute-force guessing attacks.</div>
<p> </p>
<div style="padding-left: 30px;"><strong>3) Use STRONG passwords for SIP entities.</strong>  This is probably the most important step you can take.  Don&#8217;t just concatenate two words together and suffix it with &#8220;1&#8243; - if you&#8217;ve seen how sophisticated the tools are that guess passwords, you&#8217;d understand that trivial obfuscation like that is a minor hinderance to a modern CPU.  Use symbols, numbers, and a mix of upper and lowercase letters at least 12 digits long.</div>
<p> </p>
<div style="padding-left: 30px;"><strong>4) Block your AMI manager ports.</strong>  Use &#8220;permit=&#8221; and &#8220;deny=&#8221; lines in manager.conf to reduce inbound connections to known hosts only.  Use strong passwords here, again at least 12 characters with a complex mix of symbols, numbers, and letters.</div>
<p> </p>
<div style="padding-left: 30px;"><strong>5) Allow only one or two calls at a time per SIP entity, where possible.</strong>  At the worst, limiting your exposure to toll fraud is a wise thing to do.  This also limits your exposure when legitimate password holders on your system lose control of their passphrase - writing it on the bottom of the SIP phone, for instance, which I&#8217;ve seen.</div>
<p> </p>
<div style="padding-left: 30px;"><strong>6) Make your SIP usernames different than your extensions.</strong>  While it is convenient to have extension &#8220;1234&#8243; map to SIP entry &#8220;1234&#8243; which is also SIP user &#8220;1234&#8243;, this is an easy target for attackers to guess SIP authentication names.  Use the MAC address of the device, or some sort of combination of a common phrase + extension MD5 hash (example: from a shell prompt, try &#8220;md5 -s ThePassword5000&#8243;)</div>
<p> </p>
<div style="padding-left: 30px;"><strong>7) Ensure your [default] context is secure</strong>.  Don&#8217;t allow unauthenticated callers to reach any contexts that allow toll calls.  Permit only a limited number of active calls through your default context (use the &#8220;GROUP&#8221; function as a counter.)  Prohibit unauthenticated calls entirely (if you don&#8217;t want them) by setting &#8220;allowguest=no&#8221; in the [general] part of sip.conf.</div>
<p> </p>
<div>These 7 basics will protect most people, but there are certainly other steps you can take that are more complex and reactive.  Here is a <a title="Fail2Ban recipie" href="http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk " target="_blank">fail2ban recipe</a> which might allow you to ban endpoints based on volume of requests.  There is discussion on the asterisk-user and asterisk-dev mailing lists of incorporating this type of functionality into Asterisk - let&#8217;s hear your ideas!</div>
<p> </p>
<div>
<div>If you&#8217;d like to see an example of the tools that you&#8217;re up against, see <a title="sipautohack demo video" href="http://enablesecurity.com/products/enablesecurity-voippack-sipautohack-demo/" target="_blank">this demo video</a> of an automated attack tool that does scan, guess, and crack methods via a click-and-drool interface.</div>
<div>  </div>
</div>
<div>In summary: basic security measures will protect you against the vast majority of SIP-based brute-force attacks.  Most of the SIP attackers are fools with tools - they are opportunists who see an easy way to defraud people who have not considered the costs of insecure methods.  Asterisk has some methods to prevent the most obvious attacks from succeeding at the network level, but the most effective method of protection are the administrative issues of password robustness and username obscurity. </div>
<p> </p>
<div>JT</div>
<img src="http://feeds.feedburner.com/~r/InsideTheAsterisk/~4/PBVJodflz_Y" height="1" width="1"/>]]></content:encoded>
			<wfw:commentRss>http://blogs.digium.com/2009/03/28/sip-security/feed/</wfw:commentRss>
		<feedburner:origLink>http://blogs.digium.com/2009/03/28/sip-security/</feedburner:origLink></item>
		<item>
		<title>The Rumors Of Our Death…</title>
		<link>http://feedproxy.google.com/~r/InsideTheAsterisk/~3/-UgKtILxsYE/</link>
		<comments>http://blogs.digium.com/2009/03/26/the-rumors-of-our-death/#comments</comments>
		<pubDate>Thu, 26 Mar 2009 23:55:39 +0000</pubDate>
		<dc:creator>ssokol</dc:creator>
		
		<category><![CDATA[Partners]]></category>

		<category><![CDATA[Products]]></category>

		<category><![CDATA[skype sfa sfs]]></category>

		<guid isPermaLink="false">http://blogs.digium.com/?p=3293</guid>
		<description><![CDATA[With Skype&#8217;s recent announcement of Skype For SIP there has been a great deal of pontification on the impending death of the not-yet-released Skype For Asterisk.  I&#8217;d like to take a moment to explain why Skype For SIP (SFS) does not spell the end for Skype For Asterisk (SFA), and why Skype For Asterisk is [...]]]></description>
			<content:encoded><![CDATA[<p>With Skype&#8217;s recent announcement of Skype For SIP there has been a great deal of pontification on the impending death of the not-yet-released Skype For Asterisk.  I&#8217;d like to take a moment to explain why Skype For SIP (SFS) does not spell the end for Skype For Asterisk (SFA), and why Skype For Asterisk is still in beta.</p>
<p>First, the key differences between Skype For SIP and Skype For Asterisk:</p>
<ul>
<li>SFA can handle incoming Skype calls directly from any user on the Skype network.  SFS can receive incoming calls from Skype users names that are statically mapped to a Skype name to a SIP account.</li>
<li>SFA can place calls to any user on the Skype network.  SFS cannot place calls to Skype users.</li>
<li>SFA includes support for Skype presence information.  SFS has no support for presence.</li>
<li>SFA includes buddy list management.  SFS has no buddy list management features.</li>
<li>SFA calls are encrypted from end-to-end while SFS calls are delivered to the SFS endpoint devices (PBX) as unencrypted RTP streams.</li>
<li>SFA supports multiple media codecs including G.711 aLaw and uLaw as well as G.729.  Wide-band audio will be available in a near-term revision.  SFS supports only compressed telephony-grade G.729 media streams.</li>
</ul>
<p>Ultimately, the key differentiator is that SFA is designed to allow developers to build rich applications that are deeply integrated with the Skype network, while the SFS offering is a trunking solution for legacy IP-PBX systems that support SIP.  Both have their place and neither is specifically competitive with the other.  Both Digium and Skype are dedicated to Skype For Asterisk and both are working hard to make sure that when released it exceeds expectations.</p>
<p>So now, on to the question of why Skype For Asterisk is taking so long.   There&#8217;s a lot of technology that goes into Skype and a lot of technology that goes into Asterisk, and the underpinnings of these technologies are really, really different – which has caused some portion of the delay.  Not that this is an excuse.  We&#8217;re all professionals here and after all, it’s only software.  You can make a toaster talk to a space shuttle given enough time, effort and talent.  So if you can&#8217;t excuse the technological complexities then just take a gander at the business issues.</p>
<p>Skype has spent the past five years becoming the leading global peer-to-peer consumer VoIP service and now they&#8217;re poised to do the same in the business realm.  However, the launch of commercial offerings like SFA and SFS take Skype into new territory and require some significant thought and effort.  Skype&#8217;s consumer model is based on service agreements between Skype and individual users.  This doesn&#8217;t work for business customers.  Take for example Skype names.  In the consumer world, the relationship is simple &#8212; each Skype user has a name that they select.  Those users can then add services like SkypeIn numbers and SkypeOut credits.  In the business world the model isn&#8217;t so simple.</p>
<p>Imagine, for example, if a fictitious insurance and financial services company &#8212; call them AGI &#8212; hired Bob as a salesman.  Then imagine that Bob personally registers &#8216;BadBob&#8217; as his Skype name.   Bob starts using his Skype name to let clients call him for free over the Skype network, saving AGI the toll-free charges.  Sounds good, right?  But what happens when Bob quits and goes to work for AGI&#8217;s biggest competitor?  Suddenly there&#8217;s a world of AGI customers who are calling Bob using his personal account at his new job, and there&#8217;s nothing that his former employer can do.  Ouch.</p>
<p>Enter the Skype Business Control Panel (BCP).  Skype&#8217;s Skype For Business division has launched an all-out effort to make business customers a corporate priority. Currently the BCP is simply a way to aggregate and share Skype credit.  As business services like SFA and SFS are rolled out it will take on a much more prominent role as the administrative interface for commercial customers.</p>
<p>The next update to the BCP will allow business customers to create master accounts that are owned and managed by companies rather than individuals.  With the BCP, Bob&#8217;s account would have belonged to AGI rather than to Bob, much like his AGI email account and company credit card.  If Bob went in search of greener pastures the system administrator could simply redirect his calls to another sales guy.</p>
<p>So thus the delay.  Skype and Digium have both taken a very hard look at the Skype For Asterisk model and have agreed to take the time to make it work for businesses.  This has been by far the biggest delay.  And for whatever its worth, the same delay applies to Skype For SIP.  So before you decide to declare Skype For Asterisk DOA, take a minute and consider that we&#8217;re both trying to do something new, powerful and thus-far untried in the world of business communications.  We&#8217;ll have something out just as soon as we can.  (And not a minute sooner.)</p>
<p><em><small>Update 3/27: Chris Moore from Skype&#8217;s Skype For Business unit emailed me to correct a misunderstanding regarding processing of incoming calls from SkypeIn and from Skype users.  I&#8217;ve updated this post to reflect Chris&#8217;s comments.  -S</small></em></p>
<img src="http://feeds.feedburner.com/~r/InsideTheAsterisk/~4/-UgKtILxsYE" height="1" width="1"/>]]></content:encoded>
			<wfw:commentRss>http://blogs.digium.com/2009/03/26/the-rumors-of-our-death/feed/</wfw:commentRss>
		<feedburner:origLink>http://blogs.digium.com/2009/03/26/the-rumors-of-our-death/</feedburner:origLink></item>
		<item>
		<title>Digium’s Telecom Stimulus Act</title>
		<link>http://feedproxy.google.com/~r/InsideTheAsterisk/~3/s-Is_3kdmXY/</link>
		<comments>http://blogs.digium.com/2009/03/04/digiums-telecom-stimulus-act/#comments</comments>
		<pubDate>Wed, 04 Mar 2009 15:17:54 +0000</pubDate>
		<dc:creator>jimbutler</dc:creator>
		
		<category><![CDATA[Channel]]></category>

		<category><![CDATA[Digium]]></category>

		<category><![CDATA[Switchvox]]></category>

		<category><![CDATA[promotion]]></category>

		<category><![CDATA[rebate]]></category>

		<guid isPermaLink="false">http://blogs.digium.com/?p=3173</guid>
		<description><![CDATA[Congress recently passed the 2009 American Recovery and Reinvestment Act. Did you happen to notice the incentives for businesses to invest in their phone systems?
We didn’t either.
Therefore, Digium® is launching its own worldwide Telecom Stimulus Program to help businesses
invest in their most important piece of office equipment…their phone system!
Here&#8217;s the scoop&#8230;
Buy a Switchvox® appliance with [...]]]></description>
			<content:encoded><![CDATA[<p>Congress recently passed the 2009 American Recovery and Reinvestment Act. Did you happen to notice the incentives for businesses to invest in their phone systems?</p>
<p><strong>We didn’t either.</strong></p>
<p>Therefore, Digium® is launching its own worldwide Telecom Stimulus Program to help businesses<br />
invest in their most important piece of office equipment…their phone system!</p>
<p>Here&#8217;s the scoop&#8230;</p>
<p>Buy a Switchvox® appliance with SMB software and at least 20 subscriptions between<br />
March 1st and April 30th 2009 and receive up to a $1000 rebate!</p>
<p>See the details at these links:</p>
<ul type="disc">
<li class="MsoNormal" style="background: white none repeat scroll 0%; text-align: justify;"><span style="font-size: 9pt; font-family: &quot;Verdana&quot;,&quot;sans-serif&quot;;"><a href="http://app.en25.com/e/er.aspx?s=491&amp;lid=347&amp;elq=524DE1BFD22A4AF4B11BA18F1626EA96"><span style="color: #f6772f;">Stimulus Package Program Documentation</span></a></span></li>
<li class="MsoNormal" style="background: white none repeat scroll 0%; text-align: justify;"><span style="font-size: 9pt; font-family: &quot;Verdana&quot;,&quot;sans-serif&quot;;"><a href="http://app.en25.com/e/er.aspx?s=491&amp;lid=346&amp;elq=524DE1BFD22A4AF4B11BA18F1626EA96"><span style="color: #f6772f;">Rebate Claim Form</span></a></span></li>
</ul>
<p>This really is a great deal&#8230;check it out today!</p>
<img src="http://feeds.feedburner.com/~r/InsideTheAsterisk/~4/s-Is_3kdmXY" height="1" width="1"/>]]></content:encoded>
			<wfw:commentRss>http://blogs.digium.com/2009/03/04/digiums-telecom-stimulus-act/feed/</wfw:commentRss>
		<feedburner:origLink>http://blogs.digium.com/2009/03/04/digiums-telecom-stimulus-act/</feedburner:origLink></item>
	</channel>
</rss>
