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	<title>VoIP Users Conference &#187; asterisk</title>
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	<link>http://www.voipusersconference.org</link>
	<description> Live every Friday at 12 Noon Eastern time</description>
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	<itunes:summary>This is a weekly live meeting of people all over the world who are interested in sharing knowledge and experiences about telephony over the Internet. Guests include authors, innovators, programmers and Internet personalities.
The conference is reached by phoning in using SIP, Skype or a web page widget shown on the main web site http://vuc.me</itunes:summary>
	<itunes:author>@voipusers</itunes:author>
	<itunes:explicit>clean</itunes:explicit>
	<itunes:image href="http://www.voipusersconference.org/wp-content/uploads/powerpress/_vuc300.jpg" />
	<itunes:owner>
		<itunes:name>@voipusers</itunes:name>
		<itunes:email>itunes@voipusersconference.org</itunes:email>
	</itunes:owner>
	<managingEditor>itunes@voipusersconference.org (@voipusers)</managingEditor>
	<itunes:subtitle>VOIP allows you to do almost anything with incoming and outgoing telephone lines.</itunes:subtitle>
	<itunes:keywords>telephony,communications,international,asterisk,freeswitch,freepbx,skype</itunes:keywords>
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		<title>VoIP Users Conference &#187; asterisk</title>
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		<link>http://www.voipusersconference.org</link>
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	<itunes:category text="Technology">
		<itunes:category text="Software How-To" />
		<itunes:category text="Tech News" />
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		<item>
		<title>Kevin Fleming on Digium R-Series Products</title>
		<link>http://www.voipusersconference.org/2011/vuc-friday/</link>
		<comments>http://www.voipusersconference.org/2011/vuc-friday/#comments</comments>
		<pubDate>Fri, 21 Oct 2011 13:16:12 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[analog]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[BRI]]></category>
		<category><![CDATA[digital]]></category>
		<category><![CDATA[digium]]></category>
		<category><![CDATA[failover]]></category>
		<category><![CDATA[Kevin Fleming]]></category>
		<category><![CDATA[POTS]]></category>
		<category><![CDATA[PRI]]></category>
		<category><![CDATA[R Series]]></category>
		<category><![CDATA[Redundancy]]></category>
		<category><![CDATA[T1]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=3527</guid>
		<description><![CDATA[Kevin Fleming  joins us to talk about Digium&#8216;s Redundancy Series products. Redundancy Series (aka “R-Series”) products, are a family of inexpensive, versatile appliances that enable physical layer failover of analog (POTS) and digital (T1, E1, PRI or BRI) telephony connections to Asterisk-based communications systems. The R-Series devices allows Asterisk administrators to create high-availability solutions that [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><strong><a title="Asterisk Scalable Communications Framework," href="http://www.voipusersconference.org/2010/asterisk-scalable-communications-framework/" target="_blank">Kevin Fleming </a> joins us to talk about <a title="Digium" href="http://www.digium.com" target="_blank">Digium</a>&#8216;s Redundancy Series products.</strong></p>
<p>Redundancy Series (aka “R-Series”) products, are a family of inexpensive, versatile appliances that enable physical layer failover of analog (POTS) and digital (T1, E1, PRI or BRI) telephony connections to Asterisk-based communications systems. The R-Series devices allows Asterisk administrators to create high-availability solutions that insure uninterrupted business communications, even in the event of a catastrophic hardware or software failure. R-Series appliances include all the tools necessary to build a fully redundant active/passive communications solution. If the primary server experiences a hardware or software fault, all telecom services will automatically switch to the backup server.</p>
<p>The R-Series integrates with software to mirror file systems, manages the transition of a shared IP address and monitors system health between the primary and backup. The R-Series family consists of 2 models, the analog R800 appliance and the digital R850 appliance. Each model supports eight (8) public switched telephone (PSTN) circuits. The appliances deliver PSTN circuits to each Asterisk server via Digium’s telephony interface cards. Management and failover communications between the Asterisk Servers and the R-Series appliance is done using a set of USB connections, which also provide power—saving energy costs over the lifespan of the appliance.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2011/vuc-friday/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-544841.mp3" length="13321787" type="audio/mpeg" />
			<itunes:keywords>analog,asterisk,BRI,digital,digium,failover,Kevin Fleming,POTS,PRI,R Series,Redundancy,T1</itunes:keywords>
		<itunes:subtitle>Kevin Fleming  joins us to talk about Digium&#039;s Redundancy Series products. - Redundancy Series (aka “R-Series”) products, are a family of inexpensive, versatile appliances that enable physical layer failover of analog (POTS) and digital (T1, E1,</itunes:subtitle>
		<itunes:summary>Kevin Fleming  (http://www.voipusersconference.org/2010/asterisk-scalable-communications-framework/) joins us to talk about Digium (http://www.digium.com)&#039;s Redundancy Series products.

Redundancy Series (aka “R-Series”) products, are a family of inexpensive, versatile appliances that enable physical layer failover of analog (POTS) and digital (T1, E1, PRI or BRI) telephony connections to Asterisk-based communications systems. The R-Series devices allows Asterisk administrators to create high-availability solutions that insure uninterrupted business communications, even in the event of a catastrophic hardware or software failure. R-Series appliances include all the tools necessary to build a fully redundant active/passive communications solution. If the primary server experiences a hardware or software fault, all telecom services will automatically switch to the backup server.

The R-Series integrates with software to mirror file systems, manages the transition of a shared IP address and monitors system health between the primary and backup. The R-Series family consists of 2 models, the analog R800 appliance and the digital R850 appliance. Each model supports eight (8) public switched telephone (PSTN) circuits. The appliances deliver PSTN circuits to each Asterisk server via Digium’s telephony interface cards. Management and failover communications between the Asterisk Servers and the R-Series appliance is done using a set of USB connections, which also provide power—saving energy costs over the lifespan of the appliance.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>27:41</itunes:duration>
	</item>
		<item>
		<title>Safi Systems</title>
		<link>http://www.voipusersconference.org/2011/safi-systems/</link>
		<comments>http://www.voipusersconference.org/2011/safi-systems/#comments</comments>
		<pubDate>Thu, 24 Feb 2011 23:00:37 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Eric Erickson]]></category>
		<category><![CDATA[FreeSwitch]]></category>
		<category><![CDATA[Safi Systems]]></category>
		<category><![CDATA[SafiServer]]></category>
		<category><![CDATA[SafiWrokshop]]></category>
		<category><![CDATA[Saflets]]></category>
		<category><![CDATA[Zac Wolfe]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2868</guid>
		<description><![CDATA[Zac Wolfe and Eric Erickson join us from Safi Systems about news of their latest products. SafiWorkshop is a call flow and IVR designer. With SafiWorkshop you can design, test, debug and deploy advanced call routing applications from a single, unified development environment. SafiWorkshop also includes a prompt recorder/importer for managing IVR prompts and a [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><a href="http://safisystems.com/products"><img class="size-full wp-image-2869 alignright" title="SafiSystems" src="http://www.voipusersconference.org/wp-content/uploads/2011/02/SafiSystems.gif" alt="" width="240" height="60" /></a>Zac Wolfe and Eric Erickson join us from <a title="Safi Systems" href="http://safisystems.com/" target="_blank">Safi Systems</a> about news of their latest products.</p>
<p><a title="Safi Workshop" href="http://safisystems.com/products/safiworkshop" target="_blank">SafiWorkshop</a> is a call flow and IVR designer. With SafiWorkshop you can design, test, debug  								and deploy advanced call routing applications from a single, unified development environment.  								SafiWorkshop also includes a prompt recorder/importer for managing IVR prompts and a fully  								featured SQL query editor.</p>
<p><a title="SafiServer" href="http://safisystems.com/products/safiworkshop" target="_blank">SafiServer</a> is a standalone application server that supports a broad range of telephony applications  						and serves as the platform upon which everything from call routing to complex database driven IVRs  						can be built.  SafiServer is built on enterprise level technology and has been designed from the  						ground-up with extensibility, scalability and reliability in mind. SafiServer can be run in Windows or Linux and has the flexibility to either be installed on the same 						system as your telephone switch (Asterisk or FreeSWITCH) or on a separate machine.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2011/safi-systems/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-453283.mp3" length="21434913" type="audio/mpeg" />
			<itunes:keywords>asterisk,Eric Erickson,FreeSwitch,Safi Systems,SafiServer,SafiWrokshop,Saflets,Zac Wolfe</itunes:keywords>
		<itunes:subtitle>Zac Wolfe and Eric Erickson join us from Safi Systems about news of their latest products. - SafiWorkshop is a call flow and IVR designer. With SafiWorkshop you can design, test, debug                  and deploy advanced call routing applications from a sing...</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2011/02/SafiSystems.gif)Zac Wolfe and Eric Erickson join us from Safi Systems (http://safisystems.com/) about news of their latest products.

SafiWorkshop (http://safisystems.com/products/safiworkshop) is a call flow and IVR designer. With SafiWorkshop you can design, test, debug  								and deploy advanced call routing applications from a single, unified development environment.  								SafiWorkshop also includes a prompt recorder/importer for managing IVR prompts and a fully  								featured SQL query editor.

SafiServer (http://safisystems.com/products/safiworkshop) is a standalone application server that supports a broad range of telephony applications  						and serves as the platform upon which everything from call routing to complex database driven IVRs  						can be built.  SafiServer is built on enterprise level technology and has been designed from the  						ground-up with extensibility, scalability and reliability in mind. SafiServer can be run in Windows or Linux and has the flexibility to either be installed on the same 						system as your telephone switch (Asterisk or FreeSWITCH) or on a separate machine.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>44:35</itunes:duration>
	</item>
		<item>
		<title>Suzanne Bowen of SuperTec and DIDx</title>
		<link>http://www.voipusersconference.org/2010/suzanne-bowen-supertechnologies/</link>
		<comments>http://www.voipusersconference.org/2010/suzanne-bowen-supertechnologies/#comments</comments>
		<pubDate>Thu, 02 Dec 2010 06:43:23 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[communication software]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[DIDX]]></category>
		<category><![CDATA[related companies]]></category>
		<category><![CDATA[Super Technologies]]></category>
		<category><![CDATA[supertec]]></category>
		<category><![CDATA[Suzanne Bowen]]></category>
		<category><![CDATA[technologies]]></category>
		<category><![CDATA[telcom]]></category>
		<category><![CDATA[telephone exchanges]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2665</guid>
		<description><![CDATA[Suzanne Bowen, @SuzanneBowen, VP of Super Technologies will be joining us to share a conversation on how an ex-English teacher and fitness instructor switched to a career in IP communications and social business networking and media. We&#8217;ll also be talking about one of the SuperTec companies, DIDx and the future of VoIP. (DIDx is well-known [...]]]></description>
			<content:encoded><![CDATA[<p></p><div><a href="http://www.supertec.com/"><img class="alignright size-full wp-image-2667" title="logosuper" src="http://www.voipusersconference.org/wp-content/uploads/2010/12/logosuper.png" alt="" width="180" height="130" /></a>Suzanne Bowen, <a title="Suzanne Bowen on Twitter" href="http://twitter.com/suzannebowen" target="_blank">@SuzanneBowen</a>, VP of <a title="Super Technologies" href="http://www.supertec.com/" target="_blank">Super Technologies</a> will be joining us to share a conversation on how an  ex-English teacher and fitness instructor switched to a career in IP  communications and social business networking and media. We&#8217;ll also be talking about one of the SuperTec companies, DIDx and the  future of VoIP. (<a title="DIDX" href="http://didx.net/" target="_blank">DIDx</a> is well-known on the asterisk-biz list, among other places)</div>
<div>Super  Technologies was co-founded by Suzanne and CEO Rehan Allahwala in 1999.  Since then, with an international team of developers, technical  support and salespeople, they have connected over 18,000 wholesale  telecom-related companies to buy and/or sell SIP DID and trunking. Over half of these companies are using completely open source  applications.</div>
<div>Some of the products from SuperTec companies are highlighted on <a title="SuperTec products" href="http://www.supertec.com/products/" target="_blank">this page</a>. More about Suzanne on <a title="Suzanne Bowen on Crunchbase" href="http://www.crunchbase.com/person/suzanne-bowen-2" target="_blank">Crunchbase</a> and <a title="Connect to Suzanne Bowen on Linkedin" href="http://www.linkedin.com/in/suzannebowen" target="_blank">Linkedin</a>.</div>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/suzanne-bowen-supertechnologies/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-417542.mp3" length="23191660" type="audio/mpeg" />
			<itunes:keywords>asterisk,communication,communication software,computing,DIDX,related companies,Super Technologies,supertec,Suzanne Bowen,technologies,telcom,telephone exchanges</itunes:keywords>
		<itunes:subtitle>Suzanne Bowen, @SuzanneBowen, VP of Super Technologies will be joining us to share a conversation on how an  ex-English teacher and fitness instructor switched to a career in IP  communications and social business networking and media.</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2010/12/logosuper.png)Suzanne Bowen, @SuzanneBowen (http://twitter.com/suzannebowen), VP of Super Technologies (http://www.supertec.com/) will be joining us to share a conversation on how an  ex-English teacher and fitness instructor switched to a career in IP  communications and social business networking and media. We&#039;ll also be talking about one of the SuperTec companies, DIDx and the  future of VoIP. (DIDx (http://didx.net/) is well-known on the asterisk-biz list, among other places)
Super  Technologies was co-founded by Suzanne and CEO Rehan Allahwala in 1999.  Since then, with an international team of developers, technical  support and salespeople, they have connected over 18,000 wholesale  telecom-related companies to buy and/or sell SIP DID and trunking. Over half of these companies are using completely open source  applications.
Some of the products from SuperTec companies are highlighted on this page (http://www.supertec.com/products/). More about Suzanne on Crunchbase (http://www.crunchbase.com/person/suzanne-bowen-2) and Linkedin (http://www.linkedin.com/in/suzannebowen).</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>48:15</itunes:duration>
	</item>
		<item>
		<title>Asterisk Scalable Communications Framework,</title>
		<link>http://www.voipusersconference.org/2010/asterisk-scalable-communications-framework/</link>
		<comments>http://www.voipusersconference.org/2010/asterisk-scalable-communications-framework/#comments</comments>
		<pubDate>Fri, 29 Oct 2010 10:20:05 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[communication software]]></category>
		<category><![CDATA[communication solution]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[digium]]></category>
		<category><![CDATA[extensibility]]></category>
		<category><![CDATA[fault tolerance]]></category>
		<category><![CDATA[frameworks]]></category>
		<category><![CDATA[Kevin Fleming]]></category>
		<category><![CDATA[leverage]]></category>
		<category><![CDATA[performance]]></category>
		<category><![CDATA[scalability]]></category>
		<category><![CDATA[scalable]]></category>
		<category><![CDATA[telephone exchanges]]></category>
		<category><![CDATA[Tim Panton]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2550</guid>
		<description><![CDATA[Asterisk SCF is a framework that will allow developers to move beyond voice and also create open source video and text that can be used for anything from embedded applications to enterprise and carrier solutions, according to Digium. It is delivered as a system of distributed components that can be deployed in clusters on a [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><a title="Asterisk Scalable Communications Framework" href="http://www.asterisk.org/asterisk/scf" target="_blank"><img class="alignright size-full wp-image-2551" title="asterisklogo" src="http://www.voipusersconference.org/wp-content/uploads/2010/10/asterisklogo.jpg" alt="" width="138" height="79" />Asterisk SCF</a> is a framework that will allow developers to move beyond  voice and  also create open source video and text that can be used for  anything  from embedded applications to enterprise and carrier solutions,   according to <a title="Digium" href="http://www.digium.com/" target="_blank">Digium</a>.  It is delivered as a system of distributed  components that can be  deployed in clusters on a single system or on  many systems, and will  support pretty much all real-time IP  communications including video,  wideband audio, chat and desktop  sharing.Asterisk SCF seeks to meet four requirements for businesses looking to deploy open source communications solutions:</p>
<ul>
<li>Performance</li>
<li>Scalability</li>
<li>Fault tolerance</li>
<li>Extensibility</li>
</ul>
<p>Live from Astricon (well it was when we recorded it), Kevin Fleming, director of software technologies at Digium joins us at talk about the big Digium announcement.  This discussion was made possible  by Tim Panton and Michael Graves. Kevin, Tim and Joshua Colp were in a room, surrounded by John Todd&#8217;s tie-dyed shirts.<a title="Asterisk Scalable Communications Framework" href="http://www.asterisk.org/asterisk/scf" target="_blank"><br />
</a></p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/asterisk-scalable-communications-framework/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-412493.mp3" length="47647886" type="audio/mpeg" />
			<itunes:keywords>asterisk,communication,communication software,communication solution,computing,digium,extensibility,fault tolerance,frameworks,Kevin Fleming,leverage,performance</itunes:keywords>
		<itunes:subtitle>Asterisk SCF is a framework that will allow developers to move beyond  voice and  also create open source video and text that can be used for  anything  from embedded applications to enterprise and carrier solutions,   according to Digium.</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2010/10/asterisklogo.jpg)Asterisk SCF is a framework that will allow developers to move beyond  voice and  also create open source video and text that can be used for  anything  from embedded applications to enterprise and carrier solutions,   according to Digium (http://www.digium.com/).  It is delivered as a system of distributed  components that can be  deployed in clusters on a single system or on  many systems, and will  support pretty much all real-time IP  communications including video,  wideband audio, chat and desktop  sharing.Asterisk SCF seeks to meet four requirements for businesses looking to deploy open source communications solutions:

	* Performance
	* Scalability
	* Fault tolerance
	* Extensibility

Live from Astricon (well it was when we recorded it), Kevin Fleming, director of software technologies at Digium joins us at talk about the big Digium announcement.  This discussion was made possible  by Tim Panton and Michael Graves. Kevin, Tim and Joshua Colp were in a room, surrounded by John Todd&#039;s tie-dyed shirts.
 (http://www.asterisk.org/asterisk/scf)</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:39:12</itunes:duration>
	</item>
		<item>
		<title>VoIP Abuse Project</title>
		<link>http://www.voipusersconference.org/2010/voip-abuse-project/</link>
		<comments>http://www.voipusersconference.org/2010/voip-abuse-project/#comments</comments>
		<pubDate>Fri, 01 Oct 2010 11:00:09 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[abuse]]></category>
		<category><![CDATA[accounting]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[attacker]]></category>
		<category><![CDATA[attempted]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[communication software]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[extensions]]></category>
		<category><![CDATA[honeypot]]></category>
		<category><![CDATA[mimic]]></category>
		<category><![CDATA[private branch exchange]]></category>
		<category><![CDATA[product]]></category>
		<category><![CDATA[security]]></category>
		<category><![CDATA[telephone exchanges]]></category>
		<category><![CDATA[telephony]]></category>
		<category><![CDATA[valid]]></category>
		<category><![CDATA[voice over internet protocol]]></category>
		<category><![CDATA[VoIP Security]]></category>
		<category><![CDATA[VoIPSA]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2462</guid>
		<description><![CDATA[Thanks to those who requested this on our mailing list! We have liftoff! (VoIP Security) VoIP Abuse Project . J. Oquendo &#8220;Arkeos is a program primarily used on Unix based servers running the Asterisk Open Source PBX. The application mimics a valid extension in which an attacker, after bruteforcing an account, attempts to place calls. [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><strong><a href="http://www.voipusersconference.org/wp-content/uploads/2010/09/voipsa_small.gif"><img class="alignright size-full wp-image-2464" title="voipsa_small" src="http://www.voipusersconference.org/wp-content/uploads/2010/09/voipsa_small.gif" alt="" width="118" height="28" /></a>Thanks to those who requested this on our <a title="VUC Mailing List" href="http://vuc.me/mailinglist">mailing list</a>! We have liftoff!</strong> (VoIP Security)</p>
<p><a title="Permanent Link: VoIP Abuse Project" rel="bookmark" href="http://voipsa.org/blog/2010/09/28/voip-abuse-project/">VoIP Abuse Project</a> . J. Oquendo</p>
<p>&#8220;Arkeos is a program primarily used on Unix based servers running the  Asterisk Open Source PBX. The application mimics a valid extension in  which an attacker, after bruteforcing an account, attempts to place  calls. What the attacker doesn’t know is that their calls go nowhere.  The sole purpose of allowing them to perceive they have an account is to  track them. Where they come from, what accounts they use, what VoIP  account scanners they use, what numbers they try to dial. Because of the  flexibility of Asterisk and Unix as a whole, the application does a lot  more than stated however, I will not disclose too much otherwise any  attackers reading this might catch on. Then I won’t have anyone to play  with.&#8221;</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/voip-abuse-project/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-398475.mp3" length="28211428" type="audio/mpeg" />
			<itunes:keywords>abuse,accounting,asterisk,attacker,attempted,communication,communication software,computing,extensions,honeypot,mimic,private branch exchange</itunes:keywords>
		<itunes:subtitle>Thanks to those who requested this on our mailing list! We have liftoff! (VoIP Security) - VoIP Abuse Project . J. Oquendo - &quot;Arkeos is a program primarily used on Unix based servers running the  Asterisk Open Source PBX.</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2010/09/voipsa_small.gif)Thanks to those who requested this on our mailing list (http://vuc.me/mailinglist)! We have liftoff! (VoIP Security)

VoIP Abuse Project (http://voipsa.org/blog/2010/09/28/voip-abuse-project/) . J. Oquendo

&quot;Arkeos is a program primarily used on Unix based servers running the  Asterisk Open Source PBX. The application mimics a valid extension in  which an attacker, after bruteforcing an account, attempts to place  calls. What the attacker doesn’t know is that their calls go nowhere.  The sole purpose of allowing them to perceive they have an account is to  track them. Where they come from, what accounts they use, what VoIP  account scanners they use, what numbers they try to dial. Because of the  flexibility of Asterisk and Unix as a whole, the application does a lot  more than stated however, I will not disclose too much otherwise any  attackers reading this might catch on. Then I won’t have anyone to play  with.&quot;</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>58:42</itunes:duration>
	</item>
		<item>
		<title>Introducing AGItate – Asterisk AGI on Tropo</title>
		<link>http://www.voipusersconference.org/2010/tropo-agitate-announcement/</link>
		<comments>http://www.voipusersconference.org/2010/tropo-agitate-announcement/#comments</comments>
		<pubDate>Fri, 01 Oct 2010 10:00:50 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[Adhearsion]]></category>
		<category><![CDATA[AGI]]></category>
		<category><![CDATA[applications]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Chris Mathieu]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[communication software]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[Jason Goecke]]></category>
		<category><![CDATA[pbx]]></category>
		<category><![CDATA[private branch exchange]]></category>
		<category><![CDATA[telephont]]></category>
		<category><![CDATA[Tropo]]></category>
		<category><![CDATA[Voxeo]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2470</guid>
		<description><![CDATA[From Jason Goecke&#8217;s blog post &#8211; read the whole post here on the Tropo blog. Jason made the announcement to the world about this new project on the VoIP Users Conference! &#8220;We have decided to take it a step further and created a Tropo Scripting application that turns Tropo into a giant Asterisk application platform [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><a href="http://www.voipusersconference.org/wp-content/uploads/2010/10/tropo_logo_horiz.original.png"><img class="alignright size-full wp-image-2471" title="Tropo" src="http://www.voipusersconference.org/wp-content/uploads/2010/10/tropo_logo_horiz.original.png" alt="Tropo Announces AGItate" width="250" height="41" /></a>From Jason Goecke&#8217;s blog post &#8211; read the whole post <a title="Trop Blog Post about AGItate" href="http://blog.tropo.com/2010/10/01/tropo-now-speaks-asterisk-gateway-interface-agi/" target="_blank">here on the Tropo blog</a>. Jason made the announcement to the world about this new project on the VoIP Users Conference!</p>
<p>&#8220;We have decided to take it a step further and created a <a href="https://www.tropo.com/docs/scripting" target="_blank">Tropo Scripting</a> application that turns Tropo into a giant Asterisk application platform  in the cloud.   You can now run just about any Asterisk <a href="http://www.voip-info.org/wiki/view/Asterisk+AGI" target="_blank">AGI</a> application on Tropo.</p>
<p><a href="http://tropo.com/agitate/">Tropo AGItate</a> was started on the <a href="http://www.urbandictionary.com/define.php?term=Nerd%20Bird" target="_blank">Nerd Bird</a> (good to have in-flight WiFi) from San Jose to Austin, on my way to <a href="http://www.lonestarrubyconf.com/" target="_blank">LoneStar Ruby Conference</a>. <a href="http://twitter.com/jimfreeze" target="_blank">Jim Freeze</a> – the organizer of LSRC – had recently been to <a href="http://labs.voxeo.com/2010/07/01/announcing-adhearsionconf-2010/" target="_blank">AdhearsionConf</a> in San Francisco; I wanted to be able to show something extra special  during my talk there. On that one flight, I was able to get the basics  working and show Tropo emitting <a href="http://www.voip-info.org/wiki/view/Asterisk+AGI" target="_blank">AGI</a> during my talk, just like that. (For those non-Asterisk folks out  there, AGI is an API that lets external applications connect in to  Asterisk and fully control it).&#8221;</p>
<p>Key points:</p>
<ul>
<li>Tropo can now speak AGI over TCP to any <a href="http://www.voip-info.org/wiki/view/Asterisk+FastAGI" target="_blank">FastAGI</a> server, including <a href="http://adhearsion.com/">Adhearsion</a>, <a href="http://phpagi.sourceforge.net/" target="_blank">PHPAGI</a>, <a href="http://asterisk-java.org/" target="_blank">Asterisk-Java</a>, etc.</li>
<li>A long list of Asterisk <a href="http://github.com/tropo/tropo-agi/wiki/Supported-Adhearsion-&amp;-AGI-Methods" target="_blank">commands</a> are supported, and of course all the Tropo <a href="https://www.tropo.com/docs/scripting/element_summary.htm" target="_blank">ones</a>.</li>
<li>While the script is written in Ruby, no Ruby knowledge is necessary  to use it. Just point your Tropo application to the Ruby script on <a href="http://github.com/tropo/tropo-agitate/blob/master/lib/tropo-agitate.rb" target="_blank">Github</a>, upload a <a href="http://github.com/tropo/tropo-agitate/blob/master/tropo_agi_config/tropo_agi_config.yml" target="_blank">configuration file</a> via FTP or <a href="https://www.tropo.com/docs/scripting/tropowebdav.htm" target="_blank">WebDAV</a> to your Tropo account, and you are ready.</li>
<li>Full support of Tropo <a href="https://www.tropo.com/docs/scripting/say.htm" target="_blank">Speech-Synthesis (TTS)</a> and <a href="https://www.tropo.com/docs/scripting/ask.htm" target="_blank">Speech-Recognition (ASR)</a>, but also for Asterisk <a href="http://github.com/tropo/tropo-agi/wiki/Built-In-Asterisk-Sound-Files" target="_blank">sound files</a>. Yes, tt-monkeys works!</li>
<li>Fail over to a <a href="http://www.voip-info.org/tiki-index.php?page=SIP%20URI" target="_blank">SIP URI</a>,  in case your FastAGI server does not respond. This could be to another  Tropo application, another Asterisk box, or anything that supports SIP.</li>
<li>You get all of the Tropo channels over AGI, including SMS, Instant Messaging and Twitter, all using the same application.</li>
<li>Support for custom SIP Headers in and out of Tropo.</li>
</ul>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/tropo-agitate-announcement/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-403165.mp3" length="5689483" type="audio/mpeg" />
			<itunes:keywords>Adhearsion,AGI,applications,asterisk,Chris Mathieu,communication,communication software,computing,Jason Goecke,pbx,private branch exchange,telephont</itunes:keywords>
		<itunes:subtitle>From Jason Goecke&#039;s blog post - read the whole post here on the Tropo blog. Jason made the announcement to the world about this new project on the VoIP Users Conference! - &quot;We have decided to take it a step further and created a Tropo Scripting applic...</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2010/10/tropo_logo_horiz.original.png)From Jason Goecke&#039;s blog post - read the whole post here on the Tropo blog (http://blog.tropo.com/2010/10/01/tropo-now-speaks-asterisk-gateway-interface-agi/). Jason made the announcement to the world about this new project on the VoIP Users Conference!

&quot;We have decided to take it a step further and created a Tropo Scripting (https://www.tropo.com/docs/scripting) application that turns Tropo into a giant Asterisk application platform  in the cloud.   You can now run just about any Asterisk AGI (http://www.voip-info.org/wiki/view/Asterisk+AGI) application on Tropo.

Tropo AGItate (http://tropo.com/agitate/) was started on the Nerd Bird (http://www.urbandictionary.com/define.php?term=Nerd%20Bird) (good to have in-flight WiFi) from San Jose to Austin, on my way to LoneStar Ruby Conference (http://www.lonestarrubyconf.com/). Jim Freeze (http://twitter.com/jimfreeze) – the organizer of LSRC – had recently been to AdhearsionConf (http://labs.voxeo.com/2010/07/01/announcing-adhearsionconf-2010/) in San Francisco; I wanted to be able to show something extra special  during my talk there. On that one flight, I was able to get the basics  working and show Tropo emitting AGI (http://www.voip-info.org/wiki/view/Asterisk+AGI) during my talk, just like that. (For those non-Asterisk folks out  there, AGI is an API that lets external applications connect in to  Asterisk and fully control it).&quot;

Key points:

	* Tropo can now speak AGI over TCP to any FastAGI (http://www.voip-info.org/wiki/view/Asterisk+FastAGI) server, including Adhearsion (http://adhearsion.com/), PHPAGI (http://phpagi.sourceforge.net/), Asterisk-Java (http://asterisk-java.org/), etc.
	* A long list of Asterisk commands (http://github.com/tropo/tropo-agi/wiki/Supported-Adhearsion-&amp;-AGI-Methods) are supported, and of course all the Tropo ones (https://www.tropo.com/docs/scripting/element_summary.htm).
	* While the script is written in Ruby, no Ruby knowledge is necessary  to use it. Just point your Tropo application to the Ruby script on Github (http://github.com/tropo/tropo-agitate/blob/master/lib/tropo-agitate.rb), upload a configuration file (http://github.com/tropo/tropo-agitate/blob/master/tropo_agi_config/tropo_agi_config.yml) via FTP or WebDAV (https://www.tropo.com/docs/scripting/tropowebdav.htm) to your Tropo account, and you are ready.
	* Full support of Tropo Speech-Synthesis (TTS) (https://www.tropo.com/docs/scripting/say.htm) and Speech-Recognition (ASR) (https://www.tropo.com/docs/scripting/ask.htm), but also for Asterisk sound files (http://github.com/tropo/tropo-agi/wiki/Built-In-Asterisk-Sound-Files). Yes, tt-monkeys works!
	* Fail over to a SIP URI (http://www.voip-info.org/tiki-index.php?page=SIP%20URI),  in case your FastAGI server does not respond. This could be to another  Tropo application, another Asterisk box, or anything that supports SIP.
	* You get all of the Tropo channels over AGI, including SMS, Instant Messaging and Twitter, all using the same application.
	* Support for custom SIP Headers in and out of Tropo.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>11:47</itunes:duration>
	</item>
		<item>
		<title>A VUC Special Edition Featuring Australia&#8217;s David Rowe</title>
		<link>http://www.voipusersconference.org/2010/a-vuc-special-edition-featuring-australias-david-rowe/</link>
		<comments>http://www.voipusersconference.org/2010/a-vuc-special-edition-featuring-australias-david-rowe/#comments</comments>
		<pubDate>Fri, 17 Sep 2010 17:00:33 +0000</pubDate>
		<dc:creator>mjgraves</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[CODEC2]]></category>
		<category><![CDATA[David Rowe]]></category>
		<category><![CDATA[Mesh Potato]]></category>
		<category><![CDATA[open source]]></category>
		<category><![CDATA[OSLEC]]></category>
		<category><![CDATA[village telco]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2417</guid>
		<description><![CDATA[Friday September 17th we will have a second, special edition VUC call at a special time. The guest for this hour will be David Rowe of Rowetel. David&#8217;s accomplishments in telecom are many and varied. He&#8217;s the man behind the open source hardware that is the ATCOM IP0x range of embedded Asterisk appliances. He&#8217;s one [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><img class="alignright size-full wp-image-2419" title="david_rowe_june_2009_sm" src="http://www.voipusersconference.org/wp-content/uploads/2010/09/david_rowe_june_2009_sm.jpg" alt="" width="130" height="195" />Friday September 17th we will have a second, special edition VUC call at a special time. The guest for this hour will be David Rowe of <a href="http://www.rowetel.com/blog/">Rowetel</a>.</p>
<p>David&#8217;s accomplishments in telecom are many and varied. He&#8217;s the man behind the open source hardware that is the <a href="http://www.atcom.cn/" target="_blank">ATCOM</a> <a href="http://www.atcom.cn/products_ippbx.html" target="_blank">IP0x range of embedded Asterisk appliances</a>. He&#8217;s one of the architects of the <a href="http://www.rowetel.com/blog/?page_id=454" target="_blank">&#8220;Mesh Potato&#8221;</a> hardware at the core of the <a href="http://www.villagetelco.org/" target="_blank">Village Telco Project</a> (<a href="http://www.voipusersconference.org/2009/village-telco/">Steve Song, VUC Nov 2009</a>) His efforts in the realm of software include the <a href="http://www.rowetel.com/blog/?page_id=454" target="_blank">Open Source Line Echo Canceller</a> (OSLEC) and <a href="http://www.rowetel.com/blog/?page_id=454" target="_blank">a wholly new open source, low-bitrate voice codec</a> (Codec2.)</p>
<p>David is presently just completing a trip to <a href="http://extremecom.org/" target="_blank">Extreme Com 2010</a> in Northern India. He&#8217;s recently posted some very interesting notes on his <a href="http://www.rowetel.com/blog/?p=940" target="_blank">personal blog</a>.</p>
<p>Since David resides in Adelaide, Australia this call will be at a special time which has yet to be confirmed. It will most likely be at 7am Adelaide time,  which is  5:30pm US EDT, 2:30pm PDT, 10:30pm London, and 11:30pm in Paris.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/a-vuc-special-edition-featuring-australias-david-rowe/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-398478.mp3" length="44618107" type="audio/mpeg" />
			<itunes:keywords>asterisk,CODEC2,David Rowe,Mesh Potato,open source,OSLEC,village telco</itunes:keywords>
		<itunes:subtitle>Friday September 17th we will have a second, special edition VUC call at a special time. The guest for this hour will be David Rowe of Rowetel. - David&#039;s accomplishments in telecom are many and varied. He&#039;s the man behind the open source hardware that...</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2010/09/david_rowe_june_2009_sm.jpg)Friday September 17th we will have a second, special edition VUC call at a special time. The guest for this hour will be David Rowe of Rowetel (http://www.rowetel.com/blog/).

David&#039;s accomplishments in telecom are many and varied. He&#039;s the man behind the open source hardware that is the ATCOM (http://www.atcom.cn/) IP0x range of embedded Asterisk appliances (http://www.atcom.cn/products_ippbx.html). He&#039;s one of the architects of the &quot;Mesh Potato&quot; (http://www.rowetel.com/blog/?page_id=454) hardware at the core of the Village Telco Project (http://www.villagetelco.org/) (Steve Song, VUC Nov 2009 (http://www.voipusersconference.org/2009/village-telco/)) His efforts in the realm of software include the Open Source Line Echo Canceller (http://www.rowetel.com/blog/?page_id=454) (OSLEC) and a wholly new open source, low-bitrate voice codec (http://www.rowetel.com/blog/?page_id=454) (Codec2.)

David is presently just completing a trip to Extreme Com 2010 (http://extremecom.org/) in Northern India. He&#039;s recently posted some very interesting notes on his personal blog (http://www.rowetel.com/blog/?p=940).

Since David resides in Adelaide, Australia this call will be at a special time which has yet to be confirmed. It will most likely be at 7am Adelaide time,  which is  5:30pm US EDT, 2:30pm PDT, 10:30pm London, and 11:30pm in Paris.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:01:56</itunes:duration>
	</item>
		<item>
		<title>Freeswitch Today</title>
		<link>http://www.voipusersconference.org/2010/freeswitch-today/</link>
		<comments>http://www.voipusersconference.org/2010/freeswitch-today/#comments</comments>
		<pubDate>Fri, 27 Aug 2010 11:00:57 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Brian West]]></category>
		<category><![CDATA[COmputer networks]]></category>
		<category><![CDATA[conference]]></category>
		<category><![CDATA[fax]]></category>
		<category><![CDATA[FreeSwitch]]></category>
		<category><![CDATA[Michael Collins]]></category>
		<category><![CDATA[pbx]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[telephony]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2377</guid>
		<description><![CDATA[Some intense examination of the requirement to tell people you are recording a call with the regulars while waiting for FS bridge to successfully UNmute it self. Note that ZipDX implemented the warning when you connect to the bridge if recording is on. Some reminders about *6 to toggle mute state. The origin of &#8220;Look [...]]]></description>
			<content:encoded><![CDATA[<p></p><p>Some intense examination of the requirement to tell people you are recording a call with the regulars while waiting for FS bridge to successfully UNmute it self. Note that ZipDX implemented the warning when you connect to the bridge if recording is on. Some reminders about *6 to toggle mute state.</p>
<p><a href="http://www.amazon.com/gp/product/B00009NHC0?ie=UTF8&amp;tag=resmo&amp;linkCode=as2&amp;camp=1789&amp;creative=390957&amp;creativeASIN=B00009NHC0"><img class="alignleft size-full wp-image-2394" title="thething" src="http://www.voipusersconference.org/wp-content/uploads/2010/08/thething.jpg" alt="" width="133" height="160" /></a>The origin of &#8220;<a title="The Thing from Another World" href="http://www.amazon.com/gp/product/B00009NHC0?ie=UTF8&amp;tag=resmo&amp;linkCode=as2&amp;camp=1789&amp;creative=390957&amp;creativeASIN=B00009NHC0" target="_blank"><strong>Look to your skies for a warning!</strong></a> <a title="Keep watching the Skies" href="http://www.imdb.com/title/tt0044121/" target="_blank">Keep watching the skies!</a>&#8221; Naw, I cut that part but I wanted to add the link in this post, because I didn&#8217;t get more info from FS about what to put here. Great movie, &#8220;The Thing from Another World&#8221;, featuring James Arness (&#8220;Gunsmoke&#8221;) as &#8220;The Thing&#8221;. One of the best science fiction pictures from the fifties, and one that helped define the genre.</p>
<p>From yet another world, (the <a href="http://freeswitch.org">Freeswitch</a> bridge), Pacman sounds, and then belched forth the actual meat-space aural personna of the like of Brian West, Michael Collins, Anthony Minessale, Darren Schreiber and a few less-stellar voices&#8230;</p>
<p style="text-align: center;"><a href="http://freeswitch.org"><img class="size-full wp-image-2391 aligncenter" title="fslogo" src="http://www.voipusersconference.org/wp-content/uploads/2010/08/fslogo.gif" alt="" width="303" height="74" /></a></p>
<p style="text-align: left;">From their site at Freeswitch.org: &#8220;Created in 2006 to fill the void left by proprietary  commercial solutions.  FreeSWITCH also provides a stable telephony  platform on which many telephony applications can be developed using a  wide range of free tools.</p>
<p style="text-align: left;">
FreeSWITCH was originally designed and implemented by Anthony  Minessale with the help of Brian West and Michael Jerris.  All 3 are  former developers of the popular Asterisk open source PBX.  The project  was initiated to focus on several design goals including modularity,  cross-platform support, scalability and stability. Today, many more  developers and users contribute to the project on a daily basis.&#8221;</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/freeswitch-today/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-386700.mp3" length="50871578" type="audio/mpeg" />
			<itunes:keywords>asterisk,Brian West,COmputer networks,conference,fax,FreeSwitch,Michael Collins,pbx,sip,telephony,VoIP</itunes:keywords>
		<itunes:subtitle>Some intense examination of the requirement to tell people you are recording a call with the regulars while waiting for FS bridge to successfully UNmute it self. Note that ZipDX implemented the warning when you connect to the bridge if recording is on.</itunes:subtitle>
		<itunes:summary>Some intense examination of the requirement to tell people you are recording a call with the regulars while waiting for FS bridge to successfully UNmute it self. Note that ZipDX implemented the warning when you connect to the bridge if recording is on. Some reminders about *6 to toggle mute state.

(http://www.voipusersconference.org/wp-content/uploads/2010/08/thething.jpg)The origin of &quot;Look to your skies for a warning! Keep watching the skies! (http://www.imdb.com/title/tt0044121/)&quot; Naw, I cut that part but I wanted to add the link in this post, because I didn&#039;t get more info from FS about what to put here. Great movie, &quot;The Thing from Another World&quot;, featuring James Arness (&quot;Gunsmoke&quot;) as &quot;The Thing&quot;. One of the best science fiction pictures from the fifties, and one that helped define the genre.

From yet another world, (the Freeswitch (http://freeswitch.org) bridge), Pacman sounds, and then belched forth the actual meat-space aural personna of the like of Brian West, Michael Collins, Anthony Minessale, Darren Schreiber and a few less-stellar voices...
(http://www.voipusersconference.org/wp-content/uploads/2010/08/fslogo.gif)
From their site at Freeswitch.org: &quot;Created in 2006 to fill the void left by proprietary  commercial solutions.  FreeSWITCH also provides a stable telephony  platform on which many telephony applications can be developed using a  wide range of free tools.

FreeSWITCH was originally designed and implemented by Anthony  Minessale with the help of Brian West and Michael Jerris.  All 3 are  former developers of the popular Asterisk open source PBX.  The project  was initiated to focus on several design goals including modularity,  cross-platform support, scalability and stability. Today, many more  developers and users contribute to the project on a daily basis.&quot;</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:45:59</itunes:duration>
	</item>
		<item>
		<title>New Allison Smith Collection of Funny Prompts</title>
		<link>http://www.voipusersconference.org/2010/allison-smith-funny-prompts/</link>
		<comments>http://www.voipusersconference.org/2010/allison-smith-funny-prompts/#comments</comments>
		<pubDate>Sun, 08 Aug 2010 19:16:24 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[The Rest]]></category>
		<category><![CDATA[@voiceGal]]></category>
		<category><![CDATA[allison smith]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[asterisk pbx]]></category>
		<category><![CDATA[collections]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[communication software]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[digium]]></category>
		<category><![CDATA[download]]></category>
		<category><![CDATA[free funny]]></category>
		<category><![CDATA[funny voices]]></category>
		<category><![CDATA[humor]]></category>
		<category><![CDATA[humorous]]></category>
		<category><![CDATA[IVR]]></category>
		<category><![CDATA[joey]]></category>
		<category><![CDATA[joey lindstrom]]></category>
		<category><![CDATA[lindstrom]]></category>
		<category><![CDATA[private branch exchange]]></category>
		<category><![CDATA[prompter]]></category>
		<category><![CDATA[prompts]]></category>
		<category><![CDATA[public domain]]></category>
		<category><![CDATA[recording]]></category>
		<category><![CDATA[telephone exchanges]]></category>
		<category><![CDATA[voice]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2332</guid>
		<description><![CDATA[Allison Smith (@VoiceGal) pinged me about the collection she recorded for Joey Lindstrom for use with Asterisk pbx. The full set, which Mr. Lindstrom has generously decided to place in the public domain is available for dowload free here: http://vuc.li/FunnyAllison. Joey wrote to say this was his way to give something back to the community, [...]]]></description>
			<content:encoded><![CDATA[<p></p><p>Allison Smith (<a title="@voicegal on Twitter" href="http://twitter.com/voicegal" target="_blank">@VoiceGal</a>) pinged me about the collection she recorded for Joey Lindstrom for use with Asterisk pbx. The full set, which Mr. Lindstrom has generously decided to place in the public domain is available for dowload free here:<br />
<a title="Dwonload new prompts" href="http://vuc.li/FunnyAllison" target="_blank"> http://vuc.li/FunnyAllison</a>.</p>
<p>Joey wrote to say this was his way to give something back to the community, and he&#8217;s asking Digium to put these prompts in the Asterisk distribution should they wish to do so.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/allison-smith-funny-prompts/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>ClueCon and 2600hz Project</title>
		<link>http://www.voipusersconference.org/2010/cluecon/</link>
		<comments>http://www.voipusersconference.org/2010/cluecon/#comments</comments>
		<pubDate>Fri, 06 Aug 2010 00:00:19 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[2010]]></category>
		<category><![CDATA[annuals]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[circuit]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[communication software]]></category>
		<category><![CDATA[computer networking]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[developer conference]]></category>
		<category><![CDATA[entire]]></category>
		<category><![CDATA[FreeSwitch]]></category>
		<category><![CDATA[openser]]></category>
		<category><![CDATA[OpenSIPS]]></category>
		<category><![CDATA[project]]></category>
		<category><![CDATA[projects]]></category>
		<category><![CDATA[spectra]]></category>
		<category><![CDATA[tdm]]></category>
		<category><![CDATA[telephone exchanges]]></category>
		<category><![CDATA[telephony]]></category>
		<category><![CDATA[users]]></category>
		<category><![CDATA[voice over internet protocol]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2307</guid>
		<description><![CDATA[2600hz is home to a collection of open-source telephony software that enables the use of the FreeSWITCH, Asterisk and YATE switching libraries. Initially built around the blue.box project, we aim to provide a collection of software to power your GUI, your cloud-based telephony switch and/or your monitoring and maintenance tool set. Video testing &#8211; Next [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><strong>2600hz</strong> is home to a collection of open-source telephony software that  enables the use of the FreeSWITCH, Asterisk and YATE switching  libraries. Initially built around the blue.box project, we aim to  provide a collection of software to power your GUI, your cloud-based  telephony switch and/or your monitoring and maintenance tool set.</p>
<p><strong>Video testing</strong> &#8211; Next week we&#8217;ll be doing a real video conference, watch the site for more info</p>
<p><strong>ClueCon </strong>- No one wished to comment. Whatever.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/cluecon/feed/</wfw:commentRss>
		<slash:comments>2</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-377614.mp3" length="37282376" type="audio/mpeg" />
			<itunes:keywords>2010,annuals,asterisk,circuit,communication,communication software,computer networking,computing,developer conference,entire,FreeSwitch,openser</itunes:keywords>
		<itunes:subtitle>2600hz is home to a collection of open-source telephony software that  enables the use of the FreeSWITCH, Asterisk and YATE switching  libraries. Initially built around the blue.box project, we aim to  provide a collection of software to power your GUI,</itunes:subtitle>
		<itunes:summary>2600hz is home to a collection of open-source telephony software that  enables the use of the FreeSWITCH, Asterisk and YATE switching  libraries. Initially built around the blue.box project, we aim to  provide a collection of software to power your GUI, your cloud-based  telephony switch and/or your monitoring and maintenance tool set.

Video testing - Next week we&#039;ll be doing a real video conference, watch the site for more info

ClueCon - No one wished to comment. Whatever.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:17:40</itunes:duration>
	</item>
		<item>
		<title>Adhearsion Open-source Ruby Framework for Asterisk</title>
		<link>http://www.voipusersconference.org/2010/adhearsionconf/</link>
		<comments>http://www.voipusersconference.org/2010/adhearsionconf/#comments</comments>
		<pubDate>Thu, 22 Jul 2010 23:47:26 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[Adhearsion]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[computer programming]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[discussions]]></category>
		<category><![CDATA[facilitate]]></category>
		<category><![CDATA[framework]]></category>
		<category><![CDATA[Jay Phillips]]></category>
		<category><![CDATA[john higgins]]></category>
		<category><![CDATA[open source]]></category>
		<category><![CDATA[programming]]></category>
		<category><![CDATA[Ruby]]></category>
		<category><![CDATA[ruby programming language]]></category>
		<category><![CDATA[Voxeo]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2287</guid>
		<description><![CDATA[Everyone joining the VUC this day will get a free ticket to AdhearsionConf! AdhearsionConf has been announced and registration is open. Jay Phillips, the creator of Adhearsion, will be present for two days of talks, discussions, hacking and pair programming on all things Adhearsion. Other folks will be  there sharing their innovative uses of Adhearsion. [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><strong><a href="http://labs.voxeo.com/2010/07/01/announcing-adhearsionconf-2010/?utm_source=feedburner&amp;utm_medium=feed&amp;utm_campaign=Feed%3A+VoxeoLabs+%28Voxeo+Labs%29"><img class="alignright size-full wp-image-2288" title="adhearsionconf" src="http://www.voipusersconference.org/wp-content/uploads/2010/07/adhearsionconf.jpg" alt="" width="319" height="123" /></a>Everyone joining the VUC this day will get a free ticket to  AdhearsionConf! </strong></p>
<p><a title="AdhearsionConf" href="http://labs.voxeo.com/2010/07/01/announcing-adhearsionconf-2010/?utm_source=feedburner&amp;utm_medium=feed&amp;utm_campaign=Feed%3A+VoxeoLabs+%28Voxeo+Labs%29" target="_blank">AdhearsionConf</a> has been announced and registration is open. Jay Phillips, the creator of Adhearsion, will be present for two days  of talks, discussions, hacking and pair programming on all things <a href="http://adhearsion.com/" target="_blank">Adhearsion</a>. Other folks will be  there sharing their innovative uses of  Adhearsion.</p>
<p>John Higgins, a Developer Outreach guy at Voxeo Labs joins VUC to um.. reach out to developers and get the Adhearsion word out.</p>
<p><strong>Adhearsion</strong> is a new way to write voice-enabled applications. It&#8217;s not  just an API or library — it&#8217;s a <strong>fully-featured framework</strong>,  the first of its kind, designed for maximal code reuse and  intuitiveness. The name &#8220;Adhearsion&#8221; is a combination of &#8220;adhesion&#8221; and  &#8220;hear&#8221; because Adhearsion shines best when integrating technologies with  voice.</p>
<p>Adhearsion uses the high-level <a href="http://ruby-lang.org/">Ruby  programming language</a> and has a comprehensive regression test suite.  In the age of the social web, what other technologies are more social  than voice? Adhearsion aims to help merge this so-called &#8220;Web 2.0&#8243; world  with the previously untamable telephony world.</p>
<p><strong>It&#8217;s intuitive</strong></p>
<p>Voice applications don&#8217;t have to be hard. Adhearsion abstracts  much of the complicated telephony domain, allowing you to focus on  what&#8217;s really important: the <em>business</em> domain. Take a look at  the <a href="http://adhearsion.com/examples">Examples</a> to see what we  mean.</p>
<p><strong>It&#8217;s open-source</strong></p>
<p>We take open-source software seriously. The telecom industry,  above all, needs fresh innovative talent and Adhearsion&#8217;s here to foster  it. This is one of the final major frontiers for open-source to truly  make a transformative impact and now is an exciting time to be a part of  it.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/adhearsionconf/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
		</item>
		<item>
		<title>SIPVicious and Adhearsion</title>
		<link>http://www.voipusersconference.org/2010/sipvicious-and-adhearsion/</link>
		<comments>http://www.voipusersconference.org/2010/sipvicious-and-adhearsion/#comments</comments>
		<pubDate>Fri, 25 Jun 2010 12:55:58 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[communication software]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[exposed]]></category>
		<category><![CDATA[ip pbx]]></category>
		<category><![CDATA[meaning]]></category>
		<category><![CDATA[new tools]]></category>
		<category><![CDATA[office equipment]]></category>
		<category><![CDATA[private branch exchange]]></category>
		<category><![CDATA[release]]></category>
		<category><![CDATA[Sandro Gauci]]></category>
		<category><![CDATA[scanning]]></category>
		<category><![CDATA[scans]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[SIPVicious]]></category>
		<category><![CDATA[stop]]></category>
		<category><![CDATA[system administrator]]></category>
		<category><![CDATA[technology]]></category>
		<category><![CDATA[telephone exchanges]]></category>
		<category><![CDATA[tools]]></category>
		<category><![CDATA[unauthorized]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2226</guid>
		<description><![CDATA[Both SIPVicious and Adhearsion have announcements to make. Sandro joins us to explain &#8220;svcrash&#8221;, a new addition to the tool kit to stop attacks when they happen: A new tool has been added to SIPVicious &#8211; svcrash.py. As the name implies, it crashes something &#8211; svwar.py and svcrack.py. This tool is meant to be used [...]]]></description>
			<content:encoded><![CDATA[<p></p><p>Both <a title="SIPVicious blog" href="http://blog.sipvicious.org/2010/06/how-to-crash-sipvicious-introducing.html" target="_blank">SIPVicious</a> and Adhearsion have announcements to make. Sandro joins us to explain &#8220;svcrash&#8221;, a new addition to the tool kit to stop attacks when they happen:</p>
<p><em> A new tool has been added to SIPVicious &#8211; svcrash.py. As the name  implies, it crashes something &#8211; svwar.py and svcrack.py. This tool is  meant to be used by system administrators and organizations that are  receiving unauthorized scans on their exposed IP PBX.</em></p>
<p>Ben joins us to give some details about the evolution of <a title="Adhearsion" href="http://www.adhearsion.com" target="_blank">Adhearsion</a> which has had a recent version release:</p>
<p>This release marks the continuation of the platform with some major  enhancements and bugfixes:</p>
<ul>
<li>Support for Asterisk 1.6 (continued backward compatibility with  Asterisk 1.4)</li>
<li>Added support for ActiveLDAP</li>
<li>ActiveRecord now works properly with Adhearsion components</li>
<li>Daemonizing of Adhearsion no longer truncates log files</li>
<li>Escape commands sent to Asterisk via AGI</li>
<li>Asterisk Manager Interface (AMI) events now work when daemonized</li>
<li>Various enhancements for improved Asterisk 1.6 support</li>
<li>Various bugfixes</li>
</ul>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/sipvicious-and-adhearsion/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-371249.mp3" length="21323751" type="audio/mpeg" />
			<itunes:keywords>asterisk,communication,communication software,computing,exposed,ip pbx,meaning,new tools,office equipment,private branch exchange,release,Sandro Gauci</itunes:keywords>
		<itunes:subtitle>Both SIPVicious and Adhearsion have announcements to make. Sandro joins us to explain &quot;svcrash&quot;, a new addition to the tool kit to stop attacks when they happen: A new tool has been added to SIPVicious - svcrash.py. As the name  implies,</itunes:subtitle>
		<itunes:summary>Both SIPVicious (http://blog.sipvicious.org/2010/06/how-to-crash-sipvicious-introducing.html) and Adhearsion have announcements to make. Sandro joins us to explain &quot;svcrash&quot;, a new addition to the tool kit to stop attacks when they happen:

 A new tool has been added to SIPVicious - svcrash.py. As the name  implies, it crashes something - svwar.py and svcrack.py. This tool is  meant to be used by system administrators and organizations that are  receiving unauthorized scans on their exposed IP PBX.

Ben joins us to give some details about the evolution of Adhearsion (http://www.adhearsion.com) which has had a recent version release:

This release marks the continuation of the platform with some major  enhancements and bugfixes:

	* Support for Asterisk 1.6 (continued backward compatibility with  Asterisk 1.4)
	* Added support for ActiveLDAP
	* ActiveRecord now works properly with Adhearsion components
	* Daemonizing of Adhearsion no longer truncates log files
	* Escape commands sent to Asterisk via AGI
	* Asterisk Manager Interface (AMI) events now work when daemonized
	* Various enhancements for improved Asterisk 1.6 support
	* Various bugfixes</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>44:22</itunes:duration>
	</item>
		<item>
		<title>Astricon, AMOOCON and Skype for SIP</title>
		<link>http://www.voipusersconference.org/2010/astricon-amoocon-and-skype-for-sip/</link>
		<comments>http://www.voipusersconference.org/2010/astricon-amoocon-and-skype-for-sip/#comments</comments>
		<pubDate>Sat, 29 May 2010 05:27:19 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[AMOOCON]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Astricon 2010]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[cross-platform software]]></category>
		<category><![CDATA[fallen]]></category>
		<category><![CDATA[features of skype]]></category>
		<category><![CDATA[freeware]]></category>
		<category><![CDATA[Germany]]></category>
		<category><![CDATA[John Todd]]></category>
		<category><![CDATA[next week]]></category>
		<category><![CDATA[preview]]></category>
		<category><![CDATA[recent]]></category>
		<category><![CDATA[Rostock]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[skype]]></category>
		<category><![CDATA[skype security]]></category>
		<category><![CDATA[testing]]></category>
		<category><![CDATA[voice over internet protocol]]></category>
		<category><![CDATA[web 2.0]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2154</guid>
		<description><![CDATA[John Todd gives a preview of Astricon 2010, we chat about AMOOCON in Rostock next week and about various VoIP trials and tribulations. We have a special VUC discount code for you. Enter code AC10VUC for a significant discount. We&#8217;ve been testing Skype for SIP open beta today. This allows you to connect Skype to your [...]]]></description>
			<content:encoded><![CDATA[<p></p><p>John Todd gives a preview of <a title="Astricon 2010" href="http://astricon.net" target="_blank">Astricon 2010</a>, we chat about <a title="AMOOCON 2010" href="http://amoocon.de" target="_blank">AMOOCON</a> in Rostock next week and about various VoIP trials and tribulations. We have a special VUC discount code for you. Enter code <strong>AC10VUC </strong>for a significant discount.</p>
<p>We&#8217;ve been testing <a title="Skype for SIP Open Beta" href="http://www.skype.com/intl/en-us/business/sip/overview/" target="_blank">Skype for SIP</a> open beta today. This allows you to connect  Skype to your commercial PBX, much as Skype for Asterisk allows  connection between Skype and Asterisk. Listen to the recording to learn  more about the experiences to  date.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/astricon-amoocon-and-skype-for-sip/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-361735.mp3" length="29704659" type="audio/mpeg" />
			<itunes:keywords>AMOOCON,asterisk,Astricon 2010,communication,computing,cross-platform software,fallen,features of skype,freeware,Germany,John Todd,next week</itunes:keywords>
		<itunes:subtitle>John Todd gives a preview of Astricon 2010, we chat about AMOOCON in Rostock next week and about various VoIP trials and tribulations. We have a special VUC discount code for you. Enter code AC10VUC for a significant discount. - </itunes:subtitle>
		<itunes:summary>John Todd gives a preview of Astricon 2010 (http://astricon.net), we chat about AMOOCON (http://amoocon.de) in Rostock next week and about various VoIP trials and tribulations. We have a special VUC discount code for you. Enter code AC10VUC for a significant discount.

We&#039;ve been testing Skype for SIP (http://www.skype.com/intl/en-us/business/sip/overview/) open beta today. This allows you to connect  Skype to your commercial PBX, much as Skype for Asterisk allows  connection between Skype and Asterisk. Listen to the recording to learn  more about the experiences to  date.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:01:49</itunes:duration>
	</item>
		<item>
		<title>Nerd Vittles and the Incredible PBX</title>
		<link>http://www.voipusersconference.org/2010/nerd-vittles-incredible-pbx/</link>
		<comments>http://www.voipusersconference.org/2010/nerd-vittles-incredible-pbx/#comments</comments>
		<pubDate>Fri, 28 May 2010 11:00:52 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[communication software]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[freeware]]></category>
		<category><![CDATA[Incredible PBX]]></category>
		<category><![CDATA[nerd]]></category>
		<category><![CDATA[Nerd Vittles]]></category>
		<category><![CDATA[pbx]]></category>
		<category><![CDATA[private branch exchange]]></category>
		<category><![CDATA[pstn]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[sip call]]></category>
		<category><![CDATA[skype]]></category>
		<category><![CDATA[telephone exchanges]]></category>
		<category><![CDATA[Ward Mundy]]></category>
		<category><![CDATA[worldwide]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2007</guid>
		<description><![CDATA[The Incredible PBX: Free Skype, free worldwide SIP, free U.S./Canada PSTN plus rock-solid Asterisk security using the new Zero Internet Footprint™ design. In addition to all of the Orgasmatron magic including free calling in the U.S. and Canada courtesy of Google Voice, you now get some terrific new features: randomly generated passwords for all of [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><a href="http://nerdvittles.com/?p=677"><img class="alignleft size-full wp-image-2010" title="orgasmatronV" src="http://www.voipusersconference.org/wp-content/uploads/2010/04/orgasmatronV.jpg" alt="" width="164" height="151" /></a><a href="http://nerdvittles.com/?p=677">The Incredible PBX</a>: Free Skype, free worldwide SIP, free U.S./Canada PSTN plus rock-solid Asterisk security using the new Zero Internet Footprint™ design.</p>
<p>In addition to all of the <a href="http://nerdvittles.com/?p=675">Orgasmatron</a> magic including free calling in the U.S. and Canada courtesy of Google Voice, you now get some terrific new features: randomly generated passwords for all of your extensions, free Skype support and a new backup module both of which we&#8217;ll introduce over the next few weeks. And CallerID Superfecta now is preconfigured to work out of the box with support from dozens of providers worldwide.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/nerd-vittles-incredible-pbx/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-349937.mp3" length="28260809" type="audio/mpeg" />
			<itunes:keywords>asterisk,communication software,computing,freeware,Incredible PBX,nerd,Nerd Vittles,pbx,private branch exchange,pstn,sip,sip call</itunes:keywords>
		<itunes:subtitle>The Incredible PBX: Free Skype, free worldwide SIP, free U.S./Canada PSTN plus rock-solid Asterisk security using the new Zero Internet Footprint™ design. - In addition to all of the Orgasmatron magic including free calling in the U.S.</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2010/04/orgasmatronV.jpg)The Incredible PBX (http://nerdvittles.com/?p=677): Free Skype, free worldwide SIP, free U.S./Canada PSTN plus rock-solid Asterisk security using the new Zero Internet Footprint™ design.

In addition to all of the Orgasmatron (http://nerdvittles.com/?p=675) magic including free calling in the U.S. and Canada courtesy of Google Voice, you now get some terrific new features: randomly generated passwords for all of your extensions, free Skype support and a new backup module both of which we&#039;ll introduce over the next few weeks. And CallerID Superfecta now is preconfigured to work out of the box with support from dozens of providers worldwide.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>58:49</itunes:duration>
	</item>
		<item>
		<title>KONNECT with Martin Sunstrum</title>
		<link>http://www.voipusersconference.org/2010/konnect-office/</link>
		<comments>http://www.voipusersconference.org/2010/konnect-office/#comments</comments>
		<pubDate>Thu, 20 May 2010 23:00:25 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[bases]]></category>
		<category><![CDATA[business phone]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[communication software]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[konnect]]></category>
		<category><![CDATA[office phone systems]]></category>
		<category><![CDATA[pbx]]></category>
		<category><![CDATA[pbx system]]></category>
		<category><![CDATA[private branch exchange]]></category>
		<category><![CDATA[promises]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[telephone exchanges]]></category>
		<category><![CDATA[telephony]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2014</guid>
		<description><![CDATA[Martin Sunstrum, the founder of Aksys Networks joins us for a presentation of the KONNECT line of business VoIP phones. He will give an overview of the KONNECT Business Phones and explore the tight feature integration with Asterisk-based PBX systems. Lastly, he will present the intriguing KONNECT Office Phone System, a multi-site office phone system [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><a href="http://konnectoffice.com/"><img class="alignleft size-full wp-image-2016" title="Konnect Networks Inc." src="http://www.voipusersconference.org/wp-content/uploads/2010/04/Konnect-Networks-Inc..gif" alt="" width="162" height="174" /></a>Martin Sunstrum, the founder of Aksys Networks joins us for a presentation of the KONNECT line of business VoIP  phones.</p>
<p>He will give an overview of the <a href="http://vuc.li/cTpZky" target="_blank">KONNECT  Business Phones</a> and explore  the tight feature integration with Asterisk-based PBX systems.</p>
<p>Lastly, he will present the intriguing <a href="http://vuc.li/933gUv" target="_blank">KONNECT Office Phone System</a>, a multi-site office phone system that requires no PBX  or hosted service.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/konnect-office/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-349936.mp3" length="22768810" type="audio/mpeg" />
			<itunes:keywords>asterisk,bases,business phone,communication,communication software,computing,konnect,office phone systems,pbx,pbx system,private branch exchange,promises</itunes:keywords>
		<itunes:subtitle>Martin Sunstrum, the founder of Aksys Networks joins us for a presentation of the KONNECT line of business VoIP  phones. - He will give an overview of the KONNECT  Business Phones and explore  the tight feature integration with Asterisk-based PBX syst...</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2010/04/Konnect-Networks-Inc..gif)Martin Sunstrum, the founder of Aksys Networks joins us for a presentation of the KONNECT line of business VoIP  phones.

He will give an overview of the KONNECT  Business Phones (http://vuc.li/cTpZky) and explore  the tight feature integration with Asterisk-based PBX systems.

Lastly, he will present the intriguing KONNECT Office Phone System (http://vuc.li/933gUv), a multi-site office phone system that requires no PBX  or hosted service.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>47:26</itunes:duration>
	</item>
		<item>
		<title>All About AskoziaPBX v2.0</title>
		<link>http://www.voipusersconference.org/2010/all-about-askoziapbx-v2-0/</link>
		<comments>http://www.voipusersconference.org/2010/all-about-askoziapbx-v2-0/#comments</comments>
		<pubDate>Wed, 05 May 2010 19:05:47 +0000</pubDate>
		<dc:creator>mjgraves</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[appliance]]></category>
		<category><![CDATA[AskoziaPBX]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Blackfin]]></category>
		<category><![CDATA[embedded]]></category>
		<category><![CDATA[Michael Idema]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[x86]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2094</guid>
		<description><![CDATA[This weeks guest will be Michael Iedema, leader of the AskoziaPBX Project. Michael will discuss their recent efforts, the state of the software, growth of the developer team, project road map, etc.]]></description>
			<content:encoded><![CDATA[<p></p><p><img class="alignright size-medium wp-image-2095" title="askoziapbx" src="http://www.voipusersconference.org/wp-content/uploads/2010/05/askoziapbx-300x81.png" alt="" width="300" height="81" />This weeks guest will be <a href="http://twitter.com/michael_iedema" target="_blank">Michael Iedema</a>, leader of the <a href="http://www.askozia.com/" target="_blank">AskoziaPBX</a> Project. <a href="http://www.askozia.com/" target="_blank">AskoziaPBX</a> is a lightweight embedded Linux + Asterisk distribution with a web GUI layer based upon the <a href="http://m0n0.ch/wall/" target="_blank">m0n0wall</a> framework by Miguel Kasper.</p>
<p><a href="http://www.voipusersconference.org/2009/askozia-pbx/" target="_blank">It&#8217;s almost exactly a year since  we last discussed AskoziaPBX.</a> Earlier this week they celebrated the release of v2.0 after many months of work to migrate from FreeBSD to a Linux core. Michael will  be on-hand to discuss their recent efforts, the state of the software, growth of the developer team, project road map, etc.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/all-about-askoziapbx-v2-0/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-349934.mp3" length="26624000" type="audio/mpeg" />
			<itunes:keywords>appliance,AskoziaPBX,asterisk,Blackfin,embedded,Michael Idema,sip,VoIP,x86</itunes:keywords>
		<itunes:subtitle>This weeks guest will be Michael Iedema, leader of the AskoziaPBX Project. Michael will discuss their recent efforts, the state of the software, growth of the developer team, project road map, etc.</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2010/05/askoziapbx-300x81.png)This weeks guest will be Michael Iedema (http://twitter.com/michael_iedema), leader of the AskoziaPBX (http://www.askozia.com/) Project. AskoziaPBX (http://www.askozia.com/) is a lightweight embedded Linux + Asterisk distribution with a web GUI layer based upon the m0n0wall (http://m0n0.ch/wall/) framework by Miguel Kasper.

It&#039;s almost exactly a year since  we last discussed AskoziaPBX. (http://www.voipusersconference.org/2009/askozia-pbx/) Earlier this week they celebrated the release of v2.0 after many months of work to migrate from FreeBSD to a Linux core. Michael will  be on-hand to discuss their recent efforts, the state of the software, growth of the developer team, project road map, etc.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:03:22</itunes:duration>
	</item>
		<item>
		<title>Part 2: USB, SBC, NAT</title>
		<link>http://www.voipusersconference.org/2010/usb-sbc-nat/</link>
		<comments>http://www.voipusersconference.org/2010/usb-sbc-nat/#comments</comments>
		<pubDate>Fri, 30 Apr 2010 10:00:11 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Bob Cohen]]></category>
		<category><![CDATA[broadband]]></category>
		<category><![CDATA[cellphones]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[Dave Michels]]></category>
		<category><![CDATA[David Frankel]]></category>
		<category><![CDATA[discussions]]></category>
		<category><![CDATA[freeware]]></category>
		<category><![CDATA[Future Video]]></category>
		<category><![CDATA[gateway]]></category>
		<category><![CDATA[gigaset communications]]></category>
		<category><![CDATA[Karl Fife]]></category>
		<category><![CDATA[Michael Graves]]></category>
		<category><![CDATA[Neil Fusillo]]></category>
		<category><![CDATA[network address translation]]></category>
		<category><![CDATA[online social networking]]></category>
		<category><![CDATA[Pascal Doré]]></category>
		<category><![CDATA[portable software]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[skype]]></category>
		<category><![CDATA[Tim Panton]]></category>
		<category><![CDATA[usb]]></category>
		<category><![CDATA[usb connection]]></category>
		<category><![CDATA[user conference]]></category>
		<category><![CDATA[voice over internet protocol]]></category>
		<category><![CDATA[voip user]]></category>
		<category><![CDATA[zeeek]]></category>
		<category><![CDATA[ZipDX]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=2051</guid>
		<description><![CDATA[We talk about several topics in part 2 including Skype, USB connections for VoIP, new devices like the Gigaset One, a cellphone gateway. Lots of talk about NAT with Asterisk and SIP. The usual chatter from friends around the table in our weekly &#8220;club&#8221;. Among the participants Tim @steely_glint, Neil from IdeaSIP, Karl @karlfife, Michael [...]]]></description>
			<content:encoded><![CDATA[<p></p><p>We talk about several topics in part 2 including Skype, USB connections for VoIP, new devices like the <a title="Gigaset One" href="http://www.gigasetone.com" target="_blank">Gigaset One</a>, a cellphone gateway. Lots of talk about NAT with <a title="Asterisk" href="http://asterisk.org" target="_blank">Asterisk</a> and SIP. The usual chatter from friends around the table in our weekly &#8220;club&#8221;.</p>
<p>Among the participants <a title="Tim Panton" href="http://twitter.com/steely_glint" target="_blank">Tim @steely_glint</a>, <a title="IdeaSIP" href="http://ideasip.com" target="_blank">Neil from IdeaSIP</a>, Karl <a title="Karl Fife on Twitter" href="http://twitter.com/karlfife" target="_blank">@karlfife</a>, Michael @mjgraves and <a title="Graves on SoHo VoIP" href="http://www.mgraves.org/voip/" target="_blank">Graves on SoHo VoIP</a>, David Frankel from <a title="ZipDX" href="http://www.zipdx.com" target="_blank">ZipDX</a> who furnish our great wideband conference facilities, Dave <a title="Dave Michels on Twitter" href="http://twitter.com/davemichels" target="_blank">@DaveMichels</a> from <a title="Dave Michels" href="http://www.pindropsoup.com/" target="_blank">Pin Drop Soup</a>, Pascal from <a title="Media5" href="http://media5corp.com" target="_blank">Media5</a>.</p>
<p>Welcome to a very old friend of mine, Bob from <a title="Future Video" href="http://www.futurevideo.com/" target="_blank">Future Video</a>. The VUC encourages new people to be a part of our weekly talks. Please join us so you can chime in, ask questions and generally have a great time.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/usb-sbc-nat/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-352458.mp3" length="39548820" type="audio/mpeg" />
			<itunes:keywords>asterisk,Bob Cohen,broadband,cellphones,communication,computing,Dave Michels,David Frankel,discussions,freeware,Future Video,gateway</itunes:keywords>
		<itunes:subtitle>We talk about several topics in part 2 including Skype, USB connections for VoIP, new devices like the Gigaset One, a cellphone gateway. Lots of talk about NAT with Asterisk and SIP. The usual chatter from friends around the table in our weekly &quot;club&quot;. </itunes:subtitle>
		<itunes:summary>We talk about several topics in part 2 including Skype, USB connections for VoIP, new devices like the Gigaset One (http://www.gigasetone.com), a cellphone gateway. Lots of talk about NAT with Asterisk (http://asterisk.org) and SIP. The usual chatter from friends around the table in our weekly &quot;club&quot;.

Among the participants Tim @steely_glint (http://twitter.com/steely_glint), Neil from IdeaSIP (http://ideasip.com), Karl @karlfife (http://twitter.com/karlfife), Michael @mjgraves and Graves on SoHo VoIP (http://www.mgraves.org/voip/), David Frankel from ZipDX (http://www.zipdx.com) who furnish our great wideband conference facilities, Dave @DaveMichels (http://twitter.com/davemichels) from Pin Drop Soup (http://www.pindropsoup.com/), Pascal from Media5 (http://media5corp.com).

Welcome to a very old friend of mine, Bob from Future Video (http://www.futurevideo.com/). The VUC encourages new people to be a part of our weekly talks. Please join us so you can chime in, ask questions and generally have a great time.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:22:20</itunes:duration>
	</item>
		<item>
		<title>Tim and David&#8217;s Excellent Island Telephony Adventure</title>
		<link>http://www.voipusersconference.org/2010/island-telephony-adventure/</link>
		<comments>http://www.voipusersconference.org/2010/island-telephony-adventure/#comments</comments>
		<pubDate>Fri, 16 Apr 2010 11:00:54 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[computer networking]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[David A. Burgess]]></category>
		<category><![CDATA[deploying]]></category>
		<category><![CDATA[global system for mobile communications]]></category>
		<category><![CDATA[gsm]]></category>
		<category><![CDATA[gsm phone system]]></category>
		<category><![CDATA[niue]]></category>
		<category><![CDATA[openbts]]></category>
		<category><![CDATA[projects]]></category>
		<category><![CDATA[simple network management protocol]]></category>
		<category><![CDATA[system]]></category>
		<category><![CDATA[Tim Panton]]></category>
		<category><![CDATA[universal software radio peripheral]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=1927</guid>
		<description><![CDATA[Tim Panton and David Burgess describe the project to deploy the first GSM phone system to the small island nation of Niue. The system consists of OpenSource software and (largely) commodity hardware. The project uses OpenBTS, Asterisk and USRP. Tim will try to describe (with help from OpenBTS&#8216;s David Burgess) the technical, regulatory and logistical [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><strong>Tim Panton and David Burgess describe the project to deploy the first GSM phone system to the small island nation of Niue.</strong></p>
<p>The system consists of OpenSource software and (largely) commodity hardware. The project uses OpenBTS, Asterisk and USRP. Tim will try to describe (with help from <a title="OpenBTS" href="http://ecommconf.com/blog/2009/02/david-burgess-on-openbts.html" target="_blank">OpenBTS</a>&#8216;s David Burgess) the technical, regulatory and logistical challenges that we faced before we could make the first call. See also <a title="Slide eComm 2009" href="http://www.slideshare.net/eCommConf/david-a-burgesss-presentation-at-ecomm-2009" target="_blank">David Burgess&#8217; presentation at eComm 2009</a>.</p>
<div id="attachment_1932" class="wp-caption alignleft" style="width: 364px">
	<a href="http://babyis60.wordpress.com/2010/02/25/the-island-phone-system-adventure/"><img class="size-full wp-image-1932   " title="btslastmin" src="http://www.voipusersconference.org/wp-content/uploads/2010/04/btslastmin.jpg" alt="" width="364" height="273" /></a>
	<p class="wp-caption-text">David Burgess, last minute tweaks</p>
</div>
<p>VUC regular contributor <a title="Tim's Blog Baby is 60" href="http://babyis60.wordpress.com/" target="_blank">Tim Panton</a> has been a software developer and consultant at <a title="Westhawk Ltd" href="http://www.westhawk.co.uk/" target="_blank">Westhawk Ltd</a> more than 25 years, working on a diverse range of projects, from chemical plant simulation to tourism web sites. He is also a contributor to open source projects, in particular GJTAPI a framework for implementing JTAPI (The Java Telephony API) and Westhawk&#8217;s Java SNMP stack. In recent years Tim has been predominantly involved in Asterisk development and implementation, working closely with key industry players he has been championing innovative integration of voice technologies . Tim is also the technical director of  <a title="Phone From Here" href="http://phonefromhere.com" target="_blank">PhoneFromHere.com</a> Ltd.</p>
<p><a href="http://vuc.li/8Z8Zbn">LINKS from the discussion</a>.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/island-telephony-adventure/feed/</wfw:commentRss>
		<slash:comments>3</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-334985.mp3" length="40255227" type="audio/mpeg" />
			<itunes:keywords>asterisk,communication,computer networking,computing,David A. Burgess,deploying,global system for mobile communications,gsm,gsm phone system,niue,openbts,projects</itunes:keywords>
		<itunes:subtitle>Tim Panton and David Burgess describe the project to deploy the first GSM phone system to the small island nation of Niue. - The system consists of OpenSource software and (largely) commodity hardware. The project uses OpenBTS, Asterisk and USRP.</itunes:subtitle>
		<itunes:summary>Tim Panton and David Burgess describe the project to deploy the first GSM phone system to the small island nation of Niue.

The system consists of OpenSource software and (largely) commodity hardware. The project uses OpenBTS, Asterisk and USRP. Tim will try to describe (with help from OpenBTS (http://ecommconf.com/blog/2009/02/david-burgess-on-openbts.html)&#039;s David Burgess) the technical, regulatory and logistical challenges that we faced before we could make the first call. See also David Burgess&#039; presentation at eComm 2009 (http://www.slideshare.net/eCommConf/david-a-burgesss-presentation-at-ecomm-2009).



VUC regular contributor Tim Panton (http://babyis60.wordpress.com/) has been a software developer and consultant at Westhawk Ltd (http://www.westhawk.co.uk/) more than 25 years, working on a diverse range of projects, from chemical plant simulation to tourism web sites. He is also a contributor to open source projects, in particular GJTAPI a framework for implementing JTAPI (The Java Telephony API) and Westhawk&#039;s Java SNMP stack. In recent years Tim has been predominantly involved in Asterisk development and implementation, working closely with key industry players he has been championing innovative integration of voice technologies . Tim is also the technical director of  PhoneFromHere.com (http://phonefromhere.com) Ltd.

LINKS from the discussion (http://vuc.li/8Z8Zbn).</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:23:48</itunes:duration>
	</item>
		<item>
		<title>Asterisk Reporting with Sam Reports</title>
		<link>http://www.voipusersconference.org/2010/sam-reports/</link>
		<comments>http://www.voipusersconference.org/2010/sam-reports/#comments</comments>
		<pubDate>Sat, 27 Mar 2010 11:04:40 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[asterisk solution]]></category>
		<category><![CDATA[communication]]></category>
		<category><![CDATA[communication software]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[custom software]]></category>
		<category><![CDATA[enable]]></category>
		<category><![CDATA[powerful]]></category>
		<category><![CDATA[reports]]></category>
		<category><![CDATA[rich]]></category>
		<category><![CDATA[sam]]></category>
		<category><![CDATA[telephone exchanges]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=1879</guid>
		<description><![CDATA[[audio:http://recordings.talkshoe.com/TC-22622/TS-338121.mp3&#124;titles=Sam Reports on Voipathon] Mihaela from Token d.o.o. joins us for a chat about Sam Reports, custom software from her family company located in Croatia. She and her husband have been running an Asterisk integration shop since 2007 and writing custom software to ease and enhance working with Asterisk. SAM Reports is aimed at small [...]]]></description>
			<content:encoded><![CDATA[<p></p><p style="text-align: left;">[audio:http://recordings.talkshoe.com/TC-22622/TS-338121.mp3|titles=Sam Reports on Voipathon]</p>
<p>Mihaela from <a href="http://samreports.com">Token d.o.o.</a> joins us for a chat about Sam Reports, custom software from her family company located in Croatia. She and her husband have been running an Asterisk integration shop since 2007 and writing custom software to ease and enhance working with Asterisk.</p>
<p><a href="http://samreports.com">SAM Reports</a> is aimed at small businesses using Asterisk.  I built it to solve the pain that I had : provide my clients, small businesses, with a reporting solution without the need to make any changes to their Asterisk boxes. And also to give them reporting that’s easy for them to use. I wanted to be able to offer it to both my call-center clients and those without, therefore the application handles both CDRs and queues. SAM Reports enables the end users of Asterisk solutions to have rich and powerful reports without the hassle of installing additional software.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/sam-reports/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/recordings.talkshoe.com/TC-22622/TS-338121.mp3" length="13326943" type="audio/mpeg" />
			<itunes:keywords>Voip, Asterisk,Reporting</itunes:keywords>
		<itunes:subtitle>Sam Reports Software for Asterisk</itunes:subtitle>
		<itunes:summary>SAM Reports is aimed at small businesses using Asterisk. I built it to solve the pain that I had : provide my clients, small businesses, with a reporting solution without the need to make any changes to their Asterisk boxes.</itunes:summary>
		<itunes:author>Randulo Zeeek</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>27:46</itunes:duration>
	</item>
		<item>
		<title>Kamailio 3.0, The SIP Router Project</title>
		<link>http://www.voipusersconference.org/2010/kamailio3/</link>
		<comments>http://www.voipusersconference.org/2010/kamailio3/#comments</comments>
		<pubDate>Wed, 17 Mar 2010 23:00:46 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[Alex Balashov]]></category>
		<category><![CDATA[Andrei Pelinescu-Onciul]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[combine]]></category>
		<category><![CDATA[comparison of voip software]]></category>
		<category><![CDATA[computer networking]]></category>
		<category><![CDATA[computing]]></category>
		<category><![CDATA[Daniel-Constantin Mierla]]></category>
		<category><![CDATA[Kamailio]]></category>
		<category><![CDATA[openser]]></category>
		<category><![CDATA[projects]]></category>
		<category><![CDATA[router]]></category>
		<category><![CDATA[ser]]></category>
		<category><![CDATA[session initiation protocol]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[sip express router]]></category>
		<category><![CDATA[SIP router]]></category>
		<category><![CDATA[sip server]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=1767</guid>
		<description><![CDATA[[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-317307.mp3&#124;titles=Kamailio 3.0] Guests: Daniel-Constantin Mierla, Andrei Pelinescu-Onciul and Alex Balashov (Evariste Systems) Discussion of why you would need Kamailio, who uses it, what is does. Kamailio 3.0.0, the first release that combines the features of Kamailio (OpenSER) and SIP Express Router (SER) in the same SIP server instance, and the vision of future development within [...]]]></description>
			<content:encoded><![CDATA[<p></p><p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-317307.mp3|titles=Kamailio 3.0]</p>
<p>Guests: <a title="Kamailio, Daniel-Constantin Mierla" href="http://www.kamailio.org/w/daniel-constantin-mierla/" target="_blank">Daniel-Constantin Mierla</a>, <a title="Kamailio" href="http://www.kamailio.org/w/andrei-pelinescu-onciul/" target="_blank">Andrei Pelinescu-Onciul</a> and <a title="Kamailio, Alex Balashov" href="http://www.kamailio.org/w/alex-balashov/" target="_blank">Alex Balashov</a> (<a title="Evariste Systems" href="http://evaristesys.com/" target="_blank">Evariste Systems</a>)</p>
<p>Discussion of why you would need Kamailio, who uses it, what is does.</p>
<p>Kamailio 3.0.0, the first release that combines the features of Kamailio (OpenSER) and SIP Express Router (SER) in the same SIP server instance, and the vision of future development within SIP Router project.</p>
<p>Kamailio 3.0.0:</p>
<p>- new features</p>
<p>- how to use it to deal with latest challenges in communications: SIP instant messaging, presence, and social networking</p>
<p>SIP Router</p>
<p>- project achievements since its start in 2008</p>
<p>- future development</p>
<p><a href="http://www.voipusersconference.org/wp-content/uploads/2010/03/irclinks.txt">Links from the IRC  #vuc channel</a></p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/kamailio3/feed/</wfw:commentRss>
		<slash:comments>2</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-317307.mp3" length="56897840" type="audio/mpeg" />
			<itunes:keywords>voip,Kamailio,voice,Internet,programmer,development,open source</itunes:keywords>
		<itunes:subtitle>Kamailio 3.0 release</itunes:subtitle>
		<itunes:summary>Who might need Kamailio and how and why they use it.</itunes:summary>
		<itunes:author>Randulo Zeeek</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:58:30</itunes:duration>
	</item>
		<item>
		<title>Counterpath and Digium Do It!</title>
		<link>http://www.voipusersconference.org/2010/counterpath-and-digium-do-it/</link>
		<comments>http://www.voipusersconference.org/2010/counterpath-and-digium-do-it/#comments</comments>
		<pubDate>Fri, 29 Jan 2010 01:00:52 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Bria]]></category>
		<category><![CDATA[counterpath]]></category>
		<category><![CDATA[digium]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[softphone]]></category>
		<category><![CDATA[Todd Carothers]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=1498</guid>
		<description><![CDATA[We will be welcoming Bria Product Director Todd Carothers to the VUC in a second segment on Jan 29th at 1PM EST. From this press release: CounterPath and Digium(R) Launch Asterisk-Based Multimedia Softphone Based on open standards, Bria for Asterisk combines Digium&#8217;s Asterisk, AsteriskNOW and Asterisk Business Edition telephony platforms with Bria Professional, CounterPath&#8217;s enterprise-class, [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><a href="http://www.counterpath.com"><img alt="" src="http://voipusersconference.org/wp-content/uploads/2010/01/Counterpath.jpg" title="Counterpath" class="alignleft" width="244" height="76" /></a><br />
We will be welcoming Bria Product Director Todd Carothers to the VUC in a second segment on Jan 29th at 1PM EST. </p>
<p>From this <a href="http://bit.ly/729ZP2">press release</a>:</p>
<p><strong><a href="http://www.counterpath.com/">CounterPath</a> and <a href="http://digium.com">Digium</a>(R) Launch Asterisk-Based Multimedia Softphone</strong></p>
<p>Based on open standards, Bria for Asterisk combines Digium&#8217;s Asterisk, AsteriskNOW and Asterisk Business Edition telephony platforms with Bria Professional, CounterPath&#8217;s enterprise-class, multimedia softphone. Bria for Asterisk gives users advanced telephony features, yet can be quickly set up and configured with any Asterisk deployment.</p>
<p>The VUC is officially testing Bria on Windows and OS X and we hope provide valuable feedback to Counterpath during this launch.</p>
<p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-315634.mp3|titles=Bria and Counterpath with Todd Carothers]</p>
<p>Yet another group of geeks discuss the iPad<br />
[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-318318.mp3|titles=Yet another group of geeks discuss the iPad]</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/counterpath-and-digium-do-it/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-318318.mp3" length="29515699" type="audio/mpeg" />
			<itunes:keywords>asterisk,Bria,counterpath,digium,sip,softphone,Todd Carothers</itunes:keywords>
		<itunes:subtitle>We will be welcoming Bria Product Director Todd Carothers to the VUC in a second segment on Jan 29th at 1PM EST.  - From this press release: - CounterPath and Digium(R) Launch Asterisk-Based Multimedia Softphone - Based on open standards,</itunes:subtitle>
		<itunes:summary>(http://voipusersconference.org/wp-content/uploads/2010/01/Counterpath.jpg)
We will be welcoming Bria Product Director Todd Carothers to the VUC in a second segment on Jan 29th at 1PM EST. 

From this press release (http://bit.ly/729ZP2):

CounterPath (http://www.counterpath.com/) and Digium (http://digium.com)(R) Launch Asterisk-Based Multimedia Softphone

Based on open standards, Bria for Asterisk combines Digium&#039;s Asterisk, AsteriskNOW and Asterisk Business Edition telephony platforms with Bria Professional, CounterPath&#039;s enterprise-class, multimedia softphone. Bria for Asterisk gives users advanced telephony features, yet can be quickly set up and configured with any Asterisk deployment.

The VUC is officially testing Bria on Windows and OS X and we hope provide valuable feedback to Counterpath during this launch.



Yet another group of geeks discuss the iPad</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>Astlinux</title>
		<link>http://www.voipusersconference.org/2010/astlinux/</link>
		<comments>http://www.voipusersconference.org/2010/astlinux/#comments</comments>
		<pubDate>Fri, 22 Jan 2010 10:07:35 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[astlinux]]></category>
		<category><![CDATA[Darrick Hartman]]></category>
		<category><![CDATA[FreeSwitch]]></category>
		<category><![CDATA[Lonnie Abelbeck]]></category>
		<category><![CDATA[OpenSIPS]]></category>
		<category><![CDATA[Philip Prindeville]]></category>
		<category><![CDATA[release]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=1450</guid>
		<description><![CDATA[[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-306309.mp3&#124;titles=Astlinux] AstLinux is a custom Linux distribution centered around Asterisk, the Open Source PBX. Although Asterisk remains the core focus, other VoIP applications such as OpenSIPS and FreeSWITCH are also available. AstLinux has many features that make it ideal for embedded and commerical Asterisk based solutions. On the call Friday we will discuss the newly [...]]]></description>
			<content:encoded><![CDATA[<p></p><p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-306309.mp3|titles=Astlinux]<br />
<a href="http://www.astlinux.org"><img class="aligncenter size-full wp-image-1489" title="astlinuxlogo" src="http://www.voipusersconference.org/wp-content/uploads/2009/11/astlinuxlogo.png" alt="astlinuxlogo" width="320" height="64" /></a></p>
<p><a title="Astlinux official site" href="http://www.astlinux.org" target="_blank">AstLinux</a> is a custom Linux distribution centered around Asterisk, the Open Source PBX. Although Asterisk remains the core focus, other VoIP applications such as OpenSIPS and FreeSWITCH are also available. AstLinux has many features that make it ideal for embedded and commerical Asterisk based solutions.</p>
<p>On the call Friday we will discuss the newly released 0.7.0 version of AstLinux and the number of improvements over the 0.6 series of releases.  0.7.0 will be available with either Asterisk 1.4.29 or 1.6.21 (two separate download images).</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2010/astlinux/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-306309.mp3" length="25680441" type="audio/mpeg" />
			<itunes:keywords>asterisk,astlinux,Darrick Hartman,FreeSwitch,Lonnie Abelbeck,OpenSIPS,Philip Prindeville,release</itunes:keywords>
		<itunes:subtitle>AstLinux is a custom Linux distribution centered around Asterisk, the Open Source PBX. Although Asterisk remains the core focus, other VoIP applications such as OpenSIPS and FreeSWITCH are also available. AstLinux has many features that make it ideal f...</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2009/11/astlinuxlogo.png)

AstLinux (http://www.astlinux.org) is a custom Linux distribution centered around Asterisk, the Open Source PBX. Although Asterisk remains the core focus, other VoIP applications such as OpenSIPS and FreeSWITCH are also available. AstLinux has many features that make it ideal for embedded and commerical Asterisk based solutions.

On the call Friday we will discuss the newly released 0.7.0 version of AstLinux and the number of improvements over the 0.6 series of releases.  0.7.0 will be available with either Asterisk 1.4.29 or 1.6.21 (two separate download images).</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>Kamailio, Open SER</title>
		<link>http://www.voipusersconference.org/2009/asterisk-kamaillio/</link>
		<comments>http://www.voipusersconference.org/2009/asterisk-kamaillio/#comments</comments>
		<pubDate>Thu, 17 Dec 2009 23:00:40 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[Books]]></category>
		<category><![CDATA[VoIP]]></category>
		<category><![CDATA[Alex Balashov]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Building Telephony Systems with Open SER]]></category>
		<category><![CDATA[Evariste Systems]]></category>
		<category><![CDATA[Flavio E. Goncalves]]></category>
		<category><![CDATA[Kamailio]]></category>
		<category><![CDATA[Open SER]]></category>
		<category><![CDATA[Open SIPS]]></category>
		<category><![CDATA[proxy]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=1307</guid>
		<description><![CDATA[Main event, Alex Balashov on Kamailio: [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-300113.mp3&#124;titles=Kamailio with Alex Balashov] Part Zero, pre-conference chat: [audio:http://media.blubrry.com/winelover/cloud.voipusersconference.org/vuc20091218-Part0.mp3&#124;titles=Pre-conference chat] IRC Transcript 2009-12-18 Guest Alex Balashov of Evariste Systems leads us down the path to enlightenment and understanding of this complex topic, including Kamailio, the role of Asterisk, the role of SER and media gateways, etc. - Relationship of [...]]]></description>
			<content:encoded><![CDATA[<p></p><p>Main event, Alex Balashov on Kamailio: [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-300113.mp3|titles=Kamailio with Alex Balashov]</p>
<p>Part Zero, pre-conference chat: [audio:http://media.blubrry.com/winelover/cloud.voipusersconference.org/vuc20091218-Part0.mp3|titles=Pre-conference chat]</p>
<p><a href="http://www.voipusersconference.org/wp-content/uploads/2009/12/IRC20091218.txt">IRC Transcript 2009-12-18</a></p>
<p><img class="alignleft" style="padding: 8px;" title="From LOLcat.com" src="http://lolcat.com/pics/halpchair.jpg" alt="" width="200" /></p>
<p>Guest Alex Balashov of<a href="http://evaristesys.com"> Evariste Systems</a> leads us down the path to enlightenment and understanding of this complex topic, including</p>
<p><a href="http://www.kamailio.org/">Kamailio</a>, the role of Asterisk, the role of SER and media gateways, etc.</p>
<p>- Relationship of Kamailio to OpenSER project history.<br />
- What is Kamailio/OpenSER?</p>
<p>- SIP proxy<br />
- SIP server (for certain purposes, such as registrar, presence user agent, etc.)<br />
- Common uses of Kamailio.<br />
- Service delivery platform engineering and Asterisk scaling using Kamailio.</p>
<p>- Some discussion of sip-router.org initiative.</p>
<div>
<h2>&#8220;But, wait, that&#8217;s not all!&#8221; &#8211; Free stuff!</h2>
<p><a href="http://www.packtpub.com/building-telephony-systems-with-openser/mid/071209hkjje3?utm_source=voipusersconference.org&amp;utm_medium=affiliate&amp;utm_content=blog&amp;utm_campaign=mdb_001760"><img class="alignright size-full wp-image-1352" title="PacktOpenSER" src="http://www.voipusersconference.org/wp-content/uploads/2009/10/PacktOpenSER.jpg" alt="PacktOpenSER" width="193" height="248" /></a>Two electronic copies of <a href="http://www.packtpub.com/building-telephony-systems-with-openser/mid/071209hkjje3?utm_source=voipusersconference.org&amp;utm_medium=affiliate&amp;utm_content=blog&amp;utm_campaign=mdb_001760">Building Telephony Systems with OpenSER</a> by Flavio E. Goncalves have been offered to us by <a href="http:// www.PacktPub.com">Packt Publishing</a>.</p>
<p><span>What you will learn from this book</span></p>
<blockquote>
<ul>
<li>An introduction to SIP</li>
<li>An overview of SIP Proxies</li>
<li>Compiling and installing OpenSER</li>
<li>SIP Proxy default configuration</li>
<li>Implementing MySQL support for authentication</li>
<li>Installing the user portal (SerWEB)</li>
<li>Connecting to the PSTN using a gateway</li>
<li>Sending a call to voicemail</li>
<li>NAT traversal using mediaproxy</li>
<li>Billing with Freeradius and CDRTool</li>
<li>Troubleshooting tips and tools.</li>
</ul>
</blockquote>
</div>
<p>[tweetthis]</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2009/asterisk-kamaillio/feed/</wfw:commentRss>
		<slash:comments>2</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-300113.mp3" length="43993189" type="audio/mpeg" />
			<itunes:keywords>Alex Balashov,asterisk,Building Telephony Systems with Open SER,Evariste Systems,Flavio E. Goncalves,Kamailio,Open SER,Open SIPS,proxy</itunes:keywords>
		<itunes:subtitle>Main event, Alex Balashov on Kamailio:  - Part Zero, pre-conference chat:  - IRC Transcript 2009-12-18 Guest Alex Balashov of Evariste Systems leads us down the path to enlightenment and understanding of this complex topic, including - </itunes:subtitle>
		<itunes:summary>Main event, Alex Balashov on Kamailio: 

Part Zero, pre-conference chat: 

IRC Transcript 2009-12-18 (http://www.voipusersconference.org/wp-content/uploads/2009/12/IRC20091218.txt)


(http://lolcat.com/pics/halpchair.jpg)

Guest Alex Balashov of Evariste Systems (http://evaristesys.com) leads us down the path to enlightenment and understanding of this complex topic, including

Kamailio (http://www.kamailio.org/), the role of Asterisk, the role of SER and media gateways, etc.

- Relationship of Kamailio to OpenSER project history.
- What is Kamailio/OpenSER?

- SIP proxy
- SIP server (for certain purposes, such as registrar, presence user agent, etc.)
- Common uses of Kamailio.
- Service delivery platform engineering and Asterisk scaling using Kamailio.

- Some discussion of sip-router.org initiative.

&quot;But, wait, that&#039;s not all!&quot; - Free stuff!
(http://www.voipusersconference.org/wp-content/uploads/2009/10/PacktOpenSER.jpg)Two electronic copies of Building Telephony Systems with OpenSER (http://www.packtpub.com/building-telephony-systems-with-openser/mid/071209hkjje3?utm_source=voipusersconference.org&amp;utm_medium=affiliate&amp;utm_content=blog&amp;utm_campaign=mdb_001760) by Flavio E. Goncalves have been offered to us by Packt Publishing (http:// www.PacktPub.com).

What you will learn from this book


	* An introduction to SIP
	* An overview of SIP Proxies
	* Compiling and installing OpenSER
	* SIP Proxy default configuration
	* Implementing MySQL support for authentication
	* Installing the user portal (SerWEB)
	* Connecting to the PSTN using a gateway
	* Sending a call to voicemail
	* NAT traversal using mediaproxy
	* Billing with Freeradius and CDRTool
	* Troubleshooting tips and tools.




[tweetthis]</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>Allison Smith</title>
		<link>http://www.voipusersconference.org/2009/allison-smith-2/</link>
		<comments>http://www.voipusersconference.org/2009/allison-smith-2/#comments</comments>
		<pubDate>Fri, 13 Nov 2009 16:16:51 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[allison smith]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[IVR]]></category>
		<category><![CDATA[voice over]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=1216</guid>
		<description><![CDATA[Allison is writing about what she does with her new Voicegal&#8217;s Blog. Don&#8217;t forget to follow her on Twitter: @voicegal. This Friday, Allison chatted with regulars Michael Graves and John Covici about procedures and equipment used to provide the quality work with fast turnaround that make her a renowned figure in the voice-over world. There [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><a href="http://voicegal.wordpress.com/"><img class="alignleft size-full wp-image-1217" title="ali-webcam1" src="http://www.voipusersconference.org/wp-content/uploads/2009/11/ali-webcam1.jpg" alt="ali-webcam1" width="149" height="149" /></a>Allison is writing about what she does with her new <a title="Voicegal's Blog" href="http://voicegal.wordpress.com/" target="_blank">Voicegal&#8217;s Blog</a>. Don&#8217;t forget to follow her on Twitter: <a title="Allison Smith, @voicegal" href="http://twitter.com/voicegal" target="_blank">@voicegal</a>.</p>
<p>This Friday, Allison chatted with regulars Michael Graves and John Covici about procedures and equipment used to provide the quality work with fast turnaround that make her a renowned figure in the voice-over world.</p>
<p>There was also some talk about <a title="Digium: Skype for Asterisk" href="http://www.digium.com/en/products/software/skypeforasterisk.php" target="_blank">Skype for Asterisk</a>, which <a title="PhoneFromHere.com" href="http://phonefromhere.com" target="_blank">PhoneFromHere.com</a> is using to help make it easier to get into the wideband conference bridge. My own experience on the iPod is that the audio is of excellent quality. I&#8217;ve listened on it to several concerts for long durations, it not only sounds good but seems pretty stable, considering that Skype alone is often known for dropped calls.</p>
<p>Also discussed: <a title="Graves on VoIP" href="http://www.mgraves.org/voip/2009/11/royalty-free-is-the-new-fashion-in-codecs/" target="_blank">Broadvoice releases codecs into Open Source</a></p>
<p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-285271.mp3]</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2009/allison-smith-2/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-285271.mp3" length="25621367" type="audio/mpeg" />
			<itunes:keywords>allison smith,asterisk,IVR,voice over</itunes:keywords>
		<itunes:subtitle>Allison is writing about what she does with her new Voicegal&#039;s Blog. Don&#039;t forget to follow her on Twitter: @voicegal. - This Friday, Allison chatted with regulars Michael Graves and John Covici about procedures and equipment used to provide the quali...</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2009/11/ali-webcam1.jpg)Allison is writing about what she does with her new Voicegal&#039;s Blog (http://voicegal.wordpress.com/). Don&#039;t forget to follow her on Twitter: @voicegal (http://twitter.com/voicegal).

This Friday, Allison chatted with regulars Michael Graves and John Covici about procedures and equipment used to provide the quality work with fast turnaround that make her a renowned figure in the voice-over world.

There was also some talk about Skype for Asterisk (http://www.digium.com/en/products/software/skypeforasterisk.php), which PhoneFromHere.com (http://phonefromhere.com) is using to help make it easier to get into the wideband conference bridge. My own experience on the iPod is that the audio is of excellent quality. I&#039;ve listened on it to several concerts for long durations, it not only sounds good but seems pretty stable, considering that Skype alone is often known for dropped calls.

Also discussed: Broadvoice releases codecs into Open Source (http://www.mgraves.org/voip/2009/11/royalty-free-is-the-new-fashion-in-codecs/)</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>Affordable Speech Recognition</title>
		<link>http://www.voipusersconference.org/2009/affordable-speech-recognition/</link>
		<comments>http://www.voipusersconference.org/2009/affordable-speech-recognition/#comments</comments>
		<pubDate>Fri, 02 Oct 2009 00:00:33 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[digium]]></category>
		<category><![CDATA[speech recognition]]></category>
		<category><![CDATA[Vestec]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=1031</guid>
		<description><![CDATA[Vestec Speech Engine for Asterisk® Digium has just rolled out a new Asterisk speech recognition tool-kit from a partner company called Vestec.  At $99/channel, speech recognition is down to a price point where virtually any application can be speech-enabled. Also, Fax for Asterisk, Skype for Asterisk, Astricon, mobile phones and more on part two. Part [...]]]></description>
			<content:encoded><![CDATA[<p></p><p>Vestec Speech Engine for Asterisk®</p>
<p>Digium has just rolled out a new Asterisk speech recognition tool-kit from a partner company called Vestec.  At $99/channel, speech recognition is down to a price point where virtually any application can be speech-enabled.</p>
<p>Also, Fax for Asterisk, Skype for Asterisk, Astricon, mobile phones and more on part two.</p>
<p>Part 1: [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-264000.mp3|Steve Sokol and Kashif Kahn on Vestec Speech ENgine]</p>
<p>Part 2: [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-275278.mp3|titles:iVoIP SIP for mobiles Google Voice]<br />
<a href='http://www.voipusersconference.org/wp-content/uploads/2009/09/irc-20091002.txt'>Transcript IRC-20091002</a></p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2009/affordable-speech-recognition/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-264000.mp3" length="31215542" type="audio/mpeg" />
			<itunes:keywords>speech recognition,Digium</itunes:keywords>
		<itunes:subtitle>Vestec Speech Engine for Asterisk® - Digium has just rolled out a new Asterisk speech recognition tool-kit from a partner company called Vestec.  At $99/channel, speech recognition is down to a price point where virtually any application can be speech...</itunes:subtitle>
		<itunes:summary>Vestec Speech Engine for Asterisk®

Digium has just rolled out a new Asterisk speech recognition tool-kit from a partner company called Vestec.  At $99/channel, speech recognition is down to a price point where virtually any application can be speech-enabled.

Also, Fax for Asterisk, Skype for Asterisk, Astricon, mobile phones and more on part two.

Part 1: 

Part 2: 
Transcript IRC-20091002</itunes:summary>
		<itunes:author>randulo</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:00:00</itunes:duration>
	</item>
		<item>
		<title>Nir Simionovich on EC2</title>
		<link>http://www.voipusersconference.org/2009/nir-simionovich-ec2/</link>
		<comments>http://www.voipusersconference.org/2009/nir-simionovich-ec2/#comments</comments>
		<pubDate>Fri, 19 Jun 2009 06:51:27 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[Amazon]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[cloud]]></category>
		<category><![CDATA[EC2]]></category>
		<category><![CDATA[nir simionovich]]></category>
		<category><![CDATA[Perssonas]]></category>
		<category><![CDATA[Twitter follow]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=665</guid>
		<description><![CDATA[There is a full text transcript of this hour below. Part 1: (Talkshoe) [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-224823.mp3&#124;titles=Nir Simionovich Asterisk on EC2 Instance] Part 2 (ZipDX recording served from CloudFront): [audio:http://cloud.voipusersconference.org/ZipDX20090619.mp3&#124;titles=VUC Part 2  Fridays at 12 Noon Eastern Time] We&#8217;ve tried to get people to talk about this before, but we didn&#8217;t get a deep explanation. Here&#8217;s a chance [...]]]></description>
			<content:encoded><![CDATA[<p></p><p style="text-align: left;"><strong><a title="Transcription" href="http://www.voipusersconference.org/2009/06/nir-simionovich-ec2/#more-665" target="_self">There is a full text transcript of this hour below.</a><br />
</strong></p>
<p style="text-align: left;">Part 1: (Talkshoe) [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-224823.mp3|titles=Nir Simionovich Asterisk on EC2 Instance]</p>
<p style="text-align: left;">Part 2 (ZipDX recording served from CloudFront): [audio:http://cloud.voipusersconference.org/ZipDX20090619.mp3|titles=VUC Part 2  Fridays at 12 Noon Eastern Time]</p>
<p><a href="http://www.amoocon.de/talks/27"><img class="alignright" title="Nir Simionovich AMOOCON Presentation" src="http://www.amoocon.de/assets/talks/27/img/1st_frame_medium.jpeg?1243045508" alt="" width="265" height="149" /></a></p>
<p>We&#8217;ve tried to get people to talk about this before, but we didn&#8217;t get a deep explanation. Here&#8217;s a chance to ask Nir, who did a compelling presentation at AMOOCON on Asterisk on EC2, to clear up any of those nagging questions you might have.</p>
<p>Nir&#8217;s AMOOCON presentation, <a title="AMOOCON presentation materials" href="http://www.amoocon.de/talks/27" target="_blank">Dynamic Asterisk Scalability with Amazon EC2 </a>and videos are available on the AMOOCON site. Nir&#8217;s company is <a title="Greenfield Tech" href="http://www.gftc.co.il/" target="_blank">Greenfield Tech</a>.</p>
<p>Nir is also the author of <a title="AGI/PHP Book site" href="http://www.packtpub.com/asterisk-gateway-interface-programming/book/mid/140509cq5fvb" target="_blank">Asterisk Gateway Interface 1.4 and 1.6 Programming</a></p>
<p>Jerry Shuman of <a title="Perssonas" href="http://www.perssonas.com" target="_blank">Perssonas</a> (@theagent) joined the call with some great input, too.</p>
<p><a href="http://www.voipusersconference.org/wp-content/uploads/2009/06/20090619.txt">IRC Transcript 2009-06-19</a></p>
<p><strong>Follow on Twitter</strong></p>
<p>@voipusers @e4voip @mjgraves @viperdudeuk @steely_glint @teamforrest @fredposner @asteriskbot</p>
<p><span id="more-665"></span><strong>Transcription</strong></p>
<p>Nir Simionovich: I have to admit that the first person that got me heavily into EC2 is actually here and it’s John Todd. I’ve been dealing over the past two years a lot with Asterisk and Virtualization mostly VMware, and one day on the list there was this question about Asterisk and VMware if it’s doable, what is the performance, and I actually replied to it, and John came up with the idea, “Well, there’s been some talk about EC2 and Cloud computing,” and I said, “Well, why not? Let’s give it a try and see what we can do.” As time progress and what’s really progressively fast00 was the time of the Israeli elections and I was working with a company in Israel that was doing outbound dialing to get people to come into to vote. So, we ended up building the entire infrastructure of EC2 because in order to generate as many calls as we needed to, there was simply no infrastructure big enough here in Israel to hold like 32 servers. So, we ended up opening multiple EC2 accounts and doing everything from EC2 and it worked really well and that actually sprung everything into like real rapid motion and it got me heavier and heavier into EC2.</p>
<p>Zeeek: I think the first question in most people’s mind before they know anything about EC2 and I don’t know how many people know much, I don’t know a lot. I know about S3. I’ve used it. We use S3 and CloudFront of services extensively as somebody just posted the Amoocon talks, that nearest talk is number 27 and if you’re looking in IRC, you can see that, the link, otherwise, just find the talk number 27 on Amoocon.de.</p>
<p>Anyway, Nir, the first perception of EC2, you have to wonder because for years, Asterisk has been this full thing. You’ve got a Linux box and you download it and you install it. It’s fairly easy to install if you have any experience in Linux at all, and then the thing is working and you got it on your machine and you play with it, and you know, it’s free and blah blah blah, but when you’re doing an EC2 Instance of it, I guess one of the main reasons to do this is because it’s inexpensive but it’s extensible, right? During the elections if there were thousands of thousands of calls being made, you’re not going to be able to do that on your old 386 that you threw together, right?</p>
<p>Nir Simionovich: No. That would have been impossible. The thing was that we actually needed a way to expand really fast. Now, we have an infrastructure here in Israel of servers. We have like six different tools beyond servers and because we were doing also media and outbound dialing and we’re handling a lot of capacity here in Israel, we got limited by the number of concurrent calls we’re able to per machine. So, sure we’re able to push it up to about 240 concurrent calls, 260 per server. So that wasn’t big deal but we needed more and the idea of using the EC2 just sprung up saying, “OK, let’s see what we can get out of that specific system.” So, we ended up going over there initiating what’s called the medium AMI. That’s AMI based on, I believe it’s a dual core and it has 8 gigs of RAM and 160 gigabyte of disk, and we created our own AMI image, sprung those up and each one was able to get about 120 concurrent channels on it with media and we’re really happy because we could now expand as much as we want.</p>
<p>As you said, it’s pretty easy to get Asterisk up and running on your own machine. It’s pretty easy also to get Asterisk up and running on Amazon EC2. The main problem after you get it up and running is to make sure that the work that you’ve done actually stays there because the one thing that, for me at least, is pretty annoying about Amazon EC2 is that once you bring up an instance inside EC2, if you shut it down, it will simply go away. It will get deleted. So you have to maintain your work somewhere else and going about on expanding your system with EC2 requires a fairly new skill set that most Asterisk developers currently don’t yet have, but it’s like completely mashing up the entire voice world that Asterisk is pretty much dominant in the web world where most, let’s say, IVR developers are not that accustomed to working with.</p>
<p>Fred: Hi, this is Fred. I was wondering on 260 things, you said that you’re limited to about 260 calls per server and that seems a little low for a dedicated hardware back and I was wondering what kind of setup you were using or what the limiting factor was.</p>
<p>Nir Simionovich: Well, actually the limiting factor was not Asterisk itself. It was actually the number of file descriptors we had opened on the box and the network that we had, and the boxes themselves were just dual cores. These were old IBM x335 servers and we were getting about 240, 260 concurrent calls, concurrent channels with media which were interacting with the database and external sources and the application itself was pretty heavy. So, we ended up running into a load average on the box itself around four, 4 ½ and we didn’t want to breach that one. We could have gone higher. That is for sure. That shouldn’t be a problem. Yeah, we could have gone higher to about 360, 400 but we didn’t want to. We were afraid of actually harming the quality of the call.</p>
<p>Man 3: Yeah. I’m curious if you’re using Amazon Elastic Block Store for persistent drive space for any of this architecture and if you’re noticing any sort of timing issues with it?</p>
<p>Nir Simionovich: Well, I have to admit that we hadn’t used it. We didn’t see any reason to do so but the minute I’ll do a test of that, I’ll be sure to publish those.</p>
<p>Man 3: Thanks.</p>
<p>Zeeek:  I can kind of see what EC2 is about but as far as Asterisk goes, first of all, I assume that someone has—you have the possibility now just finding instances that are already built and kind of throwing them up there? In other words, rather than starting from the beginning, someone’s already done this work, correct?</p>
<p>Nir Simionovich: That is correct. There was a lot of work done by Eric Chamberlain from Voxilla. The main issue that EC2 suffered from, the main kernel issue that each of the ES, let’s say, the standard AMI images that EC2 used was a different clock being compiled into the kernel and that caused issues with things about, let’s say, conference calls. It will just come up wrong. What Eric did is create a new image that is based off a 1,000 hertz clock which is pre-set. I think its 1,000 kilohertz, sorry. The thing was that—what Eric created was something really optimized for Asterisk to be working off Amazon EC2 and that made a lot of things in a lot of the work. It took a lot of the edge of working with Asterisk and the EC2 because it came in a pre-packaged AMI. You can just take it, you can modify it, whatever you want, and you can get your systems up and running really, really fast.</p>
<p>I know of a few companies that actually managed, like these virtual call centers. What they do is actually, they initiate small instances of EC2 according to the customers they have and what they do, they just initiate the instances as they go along. That means that if somebody right now is working and they need like four hours to do an outbound campaign or 10 hours to do an inbound campaign, then they’ll just bring out a proper EC2 image saying, “OK, work off of that,” and they will just rented you a call center per hour and it’s a completely new way of thinking about how to do call centers and how to visualize PBX systems and host PBX systems. So it brings a lot of possibilities back into the mix.</p>
<p>Zeeek: Jim is asking you, Nir, in IRC about how you distribute the load among multiple EC2 instances. It’s a good question. I’m trying to picture this stuff and it’s pretty foreign to me.</p>
<p>Nir Simionovich: Well, basically what we did, we used carriers that we’re able to distribute the load according to what we needed. We just gave them all the IP numbers, all the public IP numbers of our EC2 Instances and they actually distributed the loads for us, but we could have also easily taken, let’s say, an open sear, put it in front of the system and use that to distribute the load to each of the instances.</p>
<p>Zeeek: Other question I had was with regard to configuring. So, I don’t even understand how you access this. This looks to you like you’re going to login to it and it just looks like any box, is that it? You go with SSH into it?&#8221;</p>
<p>Nir Simionovich:  &#8230;basically, Amazon EC2 is based on Xen.  So, essentially what you&#8217;re getting is, you&#8217;re getting your own Xen domain.  Basically, imagine that, if you&#8217;re familiar with Xen, then imagine that EC2 is like this enormous Xen dom zero that operates, and whatever host, whatever AMI you issue, or you initiate, you go about and you actually create your own dom inside that, let&#8217;s say that big cloud.  Now what you get is your own virtual machine.  So, you get a machine that you can SSH into it, and work with it, and do whatever you want.  It behaves exactly like a normal machine.  Sure, you have no control, well, you have some level of control about where it is located in the world, in terms of geographically saying, “Okay, I want to be in this specific data center or in that specific data center”.  But, again, that level of control doesn&#8217;t give you a lot of ability to say, “Okay, if I initiate two, or four, or five different AMI instances”, that they will be located in the same area.  And that is the main problem with using EC2, because the normal, let&#8217;s say, coupling of servers that we are very much accustomed to when building Asterisk environments doesn&#8217;t exist there.  There is no ability to say, “Okay, if I initiate database, then the database is sitting right next to the application, and there&#8217;s no latency between those two.  So, if you do that, that&#8217;s kind of a hassle.  So, you need to go about, and say, “Okay, we need to decouple our application.  We need to decouple our information storage”, and say, “Okay, we&#8217;re going to move everything outward, into a completely new location, and have Asterisk work off of that”.  And that requires new skills.</p>
<p>Zeeek:  No question about that.</p>
<p>Nir Simionovich:  By the way, somebody on the IRC just said that Open Series is now called Open SIPS, that&#8217;s not entirely correct.  Open Series is now referred to as Camie-Camiellio[sp?], and Open SIPS is a fork of Open [unintelligible], so these are two different things.</p>
<p>Dean Collins:  Randy, it&#8217;s Dean Collins.  I just wanted to jump in.  So, are you going to set up a consultancy practice offering this type of technology, and if so, what&#8217;s the URL for your company, so people know how to get in touch with you?</p>
<p>Nir Simionovich:  Well, okay.  The easiest way to get to my website is [omitted, since it was an incorrect address].</p>
<p>Zeeek:  We&#8217;ll have to remember to post that in the&#8230;</p>
<p>Nir Simionovich:  Yeah, well actually, it&#8217;s&#8230;</p>
<p>Zeeek:  &#8230;BBC site.  Oops.  Ooh, bad URL.  Bad.</p>
<p>Nir Simionovich:  [laughs]  Yeah.  Or, if you&#8217;re able, well, because most of the audience here are English speakers, it&#8217;ll be a lot easier if you go into www.greenfieldtech.net, then you&#8217;ll get to the same place, it seems.</p>
<p>Zeeek:  How did John Todd get involved in EC2, by the way?</p>
<p>John Todd:  Because it seemed to be fairly obvious, that in order to do something really big that scaled, that EC2 is a good platform for it.  There have been, previously there have been some people doing work with EC2 doing experiments, really.  But kind of a formalized test with actual results that&#8217;s been kind of lacking.  We still don&#8217;t have really good, quantifiable tests with EC2, saying we can handle this many media channels on a particular instance, before we start to see pattern problems.  And, well, really, the whole point of my talking with Neer[sp?], actually, and a couple of other people, was to see if we could get some quantifiable results on Amazon&#8217;s network.  Because, of course, Voice Over IP is very subject to things like packet loss and jitter.  I wanted to see if Amazon&#8217;s network was going to hold up to Asterisk, or RTP, in general, operating over their fairly large infrastructure, and all, I guess, non quantifiable reports seemed to indicate that it does work just fine.  But I still am having&#8230;</p>
<p>John Todd:  I don&#8217;t have any actual numbers saying that, you know, for these three days, we tested RTP streams, and we saw this packet loss to this provider.  My goal is to get Amazon to offer, and this still may happen, we had some brief conversations with the folks at Amazon, but I&#8217;d love to see them offer a service where service providers, meaning PSTN termination providers, could interconnect with Amazon&#8217;s EC2 infrastructure at no cost, and those voice packets would be transmitted at no cost to the instance users.  In other words, don&#8217;t pay for the bits in and out of Amazon&#8217;s network if you&#8217;re using this particular service provider, who is bearing the burden of network interconnect.  That would make this much more appealing for VoIP services, and application providers to outsource their activity to Amazon, and I think that that&#8217;s really just waiting for kind of a critical mass of EC2 infrastructure stuff to happen&#8230;</p>
<p>Zeeek:  And of course, you were channeling Don Quixote, as always, John.</p>
<p>John Todd:  Yeah, to some degree.  But actually, Amazon is pretty clueful, if they see this as a way to increase the number of hours people use the machines, then I don&#8217;t think that, you know, their network costs, I think, are pretty close to break even, unless you&#8217;re doing something like video.  But I think anything you&#8217;ve got to do to encourage people to get onto their system, they would be interested in.  One of the big services that is based on EC2 and Asterisk is Toolio[sp?].  All of their infrastructure right now, from what I&#8217;ve understood them to say, is that it is running on EC2, scalably, with Asterisk as the back end.  And, as a matter of fact, they&#8217;re going to be giving a talk about that at AstriCon this fall, in October.</p>
<p>Zeeek:  There was a question about, and a very good question from Jim, about bandwidth cost calculations.  Now, you did give us some idea, Neer[sp?], during your presentation about the costs.  Why don&#8217;t you run over that a little?  Whatever you can, or want to reveal about the cost of the campaign, some of the numbers.</p>
<p>Nir Simionovich:  Well, the calculation is based on&#8230;well, okay.  Let&#8217;s start from square one.  Amazon EC2 economics, the way they calculate your usage is a little different.  That means that they calculate your usage according to your instances.  For example John, I&#8217;ll just punch it up for a second.  For example, they would go about and say, “Okay, for every gigabyte that you consume, up to a certain level, you will pay something like, let&#8217;s say a cent”.  Now, if you know that specific calculation, and you&#8217;re able able to calculate what is your projected traffic, and you can calculate you&#8217;re projected traffic according to the codecs you use, and how many concurring calls you have, and how many sessions you&#8230;and how much time you will be operating the system, you&#8217;re pretty much easy to say, “Okay, I can go about and say, okay, I&#8217;ll be utilizing, let&#8217;s say 800 gigabytes over the next two weeks of traffic”.  So, you&#8217;re able to say, “Okay, 800 gigabytes of traffic, multiplied by one cent, that&#8217;s eight dollars worth of traffic”.  So, that&#8217;s how the math actually works.  Now, to say that there is a calculator that enables you to calculate that exactly, it&#8217;s a little hard to say.  I don&#8217;t think that one exists, but the math, in itself, isn&#8217;t that complex to do.  In general, just use any type of bandwidth calculator that you&#8217;ll be using for any VoIP application, and just calculate your, let&#8217;s say, megabit per second ratio, and once you have that, just multiply that by the number of seconds you have over a course of your operation, and you&#8217;ll have your number.</p>
<p>Zeeek:  It&#8217;s interesting, Dave EG[sp?], from EGW, we have a customer who uses EC2 for their telephony apps, multiple Asterisk instances, front ended by Open SIPS, you have some logic which brings up and down additional Asterisk instances for peak load times, which is basically, Neer[sp?], what you did during that election campaign.  Interesting.</p>
<p>Nir Simionovich:  That&#8217;s exactly the same.  Same idea.</p>
<p>Zeeek:  Yeah.  No question, this is great for things that you pointed out, and one of those was, hey, you&#8217;ve got this project that lasts, what was that, three days, is that right?  Something like that?</p>
<p>Nir Simionovich:  It was three days worth of dial out, and it was operating about ten to twelve hours every day.  So, we got out about 36 hours, 38 hours worth of dial out.</p>
<p>Zeeek:  It would be incredibly hard to put together a hardware platform, and stupid [laughs], to put together a hardware platform for three days, unless, you know, you worked in a hardware store, or something, to do this project of over a couple of days.  Even if it was a few weeks, whereas this is one of the things where&#8230;</p>
<p>Nir Simionovich:  That&#8217;s exactly the case.</p>
<p>Zeeek:  The whole EC2 thing shines.</p>
<p>Nir: This is exactly what it pays. Awhile back, John and I did some mass going about and saying, trying to understand if there is a validity to actually going about and running a host PBX on top of Amazon EC2, and we realized that if you operate for 24 hours, a full system is operating 24 hours, then there is no real usage for Amazon EC2 as is. If you’re a business which is doing a hosted environment, then Amazon EC2 is the best you’ll find, really. It gives you that immediate quantifiable capacity that is really, really fast,but if you’re on business, let’s say, if you’re in an office and needed a PBX system, then Amazon EC2 isn’t really the thing for you because you can get an equivalent server with inside dedicated hosting facility today for like $59 a month and that one already has 16 gigabytes of traffic with it, maintenance, managed hosting and so on and so on. So, some services make sense working off of EC2, some don’t. If you’re going to say, “OK, we want to move into the Cloud,” that specific move has to be calculated and has to be tested, let’s say, economic wise.</p>
<p>Zeeek: The agent has posted—now, I’ve seen calculators for S3 before and this is it, right? Well, I’m going to read this because I don’t have time to make a short URL, Calculator.S3.AmazonEWS.com/calc5.html.</p>
<p>Nir: That will give some insights as to how much money you’ll need to pay.</p>
<p>Zeeek: I think it’s worth trying a project, maybe messing with it to see. So once you’ve got your instance ready to go, I mean, if the elections are held next weekend, are you ready to bring that same instance up? What’s involved in bringing that back to life? Can it be done?</p>
<p>Nir: Yeah, it’ll take about five minutes.</p>
<p>Zeeek: So where is this thing stored? I mean, it’s in your account and you just click, you just check a box and click submit and you can bring it back up or is it something locally stored somewhere with you or what?</p>
<p>Nir: It’s exactly how you described. What you do is, once you build your own box, you can actually create your own AMI image that contains all your software. That’s exactly what we did. Once we had our own AMI image, the private image, we can then store inside S3 and what will happen is, the minute we may want it, we’ll just go into our AWS account and we’ll just say, “OK, we need now five instances of this specific AMI running,” and we’ll just click that and punch it up immediately. No big deal.</p>
<p>Zeeek: Interesting. The agent is talking about the ElasticFox extension and start playing with EC2. I have absolutely no notion of what that is but hopefully, either you do or he’ll call in and tell us about it.</p>
<p>Nir: Yeah. ElasticFox is a Firefox extension that enables you to play with Amazon EC2 instead of working with the web interface. It’s only for Firefox, that’s one. I think a combination of both ElasticFox and the web interface as is because the new web interfaces they’ve created which is currently and still in beta is, in my view, just a bit easier to use than the ElasticFox, just a bit.</p>
<p>Zeeek: OK, and as I mentioned, IRC, I use a Firefox plug-in sometimes for S3 or for CloudFront. So, I assume it’s a similar concept. You just kind of dumps it down.</p>
<p>Nir: Yeah, similar concept. Yeah.</p>
<p>Zeeek: It’s kind of amazing to me that—and this goes back to what we were saying with John Todd that there’s no good reason to think that Amazon’s network would allow for real time. The CloudFront thing, the idea there is that you can be serving video and stuff like that and that it will push it out to the edge servers that will be the most efficient. On the other hand, I’m not sure how that translates to VoIP and I’m surprised to learn that they’re able to do the real time. Who knew, you know. That’s the way I feel about it. So it’s kind of surprising that they have the capacity—you don’t have any problem with call quality apparently, right?</p>
<p>Nir: No. Well, let’s say this, as long as you’re working with carriers which were interconnected with, let’s say…</p>
<p>[Music playing]</p>
<p>Zeeek: Yeah. Let me try to find the source of that.</p>
<p>Nir: OK.</p>
<p>Zeeek: That may have been Digium. Hello, Digium. Thank you. They may not. Sorry.</p>
<p>Nir: That was a good one.</p>
<p>Zeeek: Yeah. Well, that’s the Asterisk built-in…</p>
<p>Nir: We can talk over the music.</p>
<p>Zeeek: Frankly, I think that was somebody in Huntsville or else I just happened to click it by accident. Sorry. That happens from time to time.</p>
<p>Nir: Yeah.</p>
<p>Zeeek: Where were we?</p>
<p>Nir: In any case—where were we? Yeah, in any case, we were working with three or four different providers in the U.S. which are considered somewhere around, let’s say the tier 2. Then, usually, these carriers are interconnected on the backbone with links that are very, very efficient and reliable over to Amazon EC2 as long as you’re hosting your instances in the U.S. So, the end result that were, let’s say one of the carriers that we were using was called TSG Global and we were getting about 24 millisecond roundtrip from Amazon EC2 to TSG Global. Now, you’ll run voice over IP on that and will have perfect quality always.</p>
<p>Zeeek: I’m trying of course the agent to call in, trying to figure out because it would be more interesting to have a discussion here than reading the stuff in IRC.</p>
<p>Nir: Yeah, absolutely.</p>
<p>Zeeek: OK. My arguments have won over. He’s going to call. He or she, you never know, right?</p>
<p>Nir: Well, he brought up a very interesting point of how to bundle your own AMI and this is actually what I was talking about, creating your own AMI image. It’s not that simple. I have to be honest. It took me about four or five hours to actually understand how to do it right and it never works from the first try. It’s really annoying. What happens is that you need to take your running machine, your running AMI and you’re creating a raw06:25 image of the hard drive, of the virtual hard drive and then you upload that into S3. That’s complicated to do but if done right it will work on the second or third try. It never works on the first try, never.</p>
<p>Zeeek: OK and we have a bleep from Southwest British Columbia. Can you hear us? Could be the agent, you never know.</p>
<p>Nir: Actually, I’d love to hear about what he just wrote on Openfire XMPP and Red5 Flash Server for video conferencing. That is really interesting because I’ve been trying to play around with Red5 but I have to admit that Flash isn’t really my thing. So, I’d love to hear about that.</p>
<p>Zeeek: OK, the agent is coming out of the wild of Southern California. Who is in the Southwest British Columbia? Say hello. Come on, don’t be shy. We’re waiting for the agent to call in. We have time.</p>
<p>Man 2: What do you do for your logs then here, just logs if you’re just bringing up these instances and using for so many hours and shutting them down? Are you uploading those at some point, back up to S3 or do you just not care about them?</p>
<p>Nir: Well, the application as is doesn’t really carry that much logging into the actual server itself but I have to admit that I never really needed to go about and offload the logs anywhere because we don’t really are about those. CDR’s are kept completely separated from the system but you could offload those into a different machine or you can, before shutting down the system, go back and offload those out of EC2. It shouldn’t be that much of a problem. Or if you really want to, you can use the—it’s not S3—what was that again—it’s called…</p>
<p>Zeeek: CloudFront?</p>
<p>Nir: No, not CloudFront. They’ve got servers where you can actually map a storage device to your AMI and then you can share&#8230;&#8221;</p>
<p>Nir Simionovich: This is exactly what it pays. Awhile back, John and I did some mass going about and saying, trying to understand if there is a validity to actually going about and running a host PBX on top of Amazon EC2, and we realized that if you operate for 24 hours, a full system is operating 24 hours, then there is no real usage for Amazon EC2 as is. If you’re a business which is doing a hosted environment, then Amazon EC2 is the best you’ll find, really. It gives you that immediate quantifiable capacity that is really, really fast,but if you’re on business, let’s say, if you’re in an office and needed a PBX system, then Amazon EC2 isn’t really the thing for you because you can get an equivalent server with inside dedicated hosting facility today for like $59 a month and that one already has 16 gigabytes of traffic with it, maintenance, managed hosting and so on and so on. So, some services make sense working off of EC2, some don’t. If you’re going to say, “OK, we want to move into the Cloud,” that specific move has to be calculated and has to be tested, let’s say, economic wise. Moderator: The agent has posted—now, I’ve seen calculators for S3 before and this is it, right? Well, I’m going to read this because I don’t have time to make a short URL, Calculator.S3.AmazonEWS.com/calc5.html. Nir Simionovich: That will give some insights as to how much money you’ll need to pay. Moderator: I think it’s worth trying a project, maybe messing with it to see. So once you’ve got your instance ready to go, I mean, if the elections are held next weekend, are you ready to bring that same instance up? What’s involved in bringing that back to life? Can it be done? Nir Simionovich: Yeah, it’ll take about five minutes. Moderator: So where is this thing stored? I mean, it’s in your account and you just click, you just check a box and click submit and you can bring it back up or is it something locally stored somewhere with you or what? Nir Simionovich: It’s exactly how you described. What you do is, once you build your own box, you can actually create your own AMI image that contains all your software. That’s exactly what we did. Once we had our own AMI image, the private image, we can then store inside S3 and what will happen is, the minute we may want it, we’ll just go into our AWS account and we’ll just say, “OK, we need now five instances of this specific AMI running,” and we’ll just click that and punch it up immediately. No big deal. Moderator: Interesting. The agent is talking about the ElasticFox extension and start playing with EC2. I have absolutely no notion of what that is but hopefully, either you do or he’ll call in and tell us about it. Nir Simionovich: Yeah. ElasticFox is a Firefox extension that enables you to play with Amazon EC2 instead of working with the web interface. It’s only for Firefox, that’s one. I think a combination of both ElasticFox and the web interface as is because the new web interfaces they’ve created which is currently and still in beta is, in my view, just a bit easier to use than the ElasticFox, just a bit. Moderator: OK, and as I mentioned, IRC, I use a Firefox plug-in sometimes for S3 or for CloudFront. So, I assume it’s a similar concept. You just kind of dumps it down. Nir Simionovich: Yeah, similar concept. Yeah. Moderator: It’s kind of amazing to me that—and this goes back to what we were saying with John Todd that there’s no good reason to think that Amazon’s network would allow for real time. The CloudFront thing, the idea there is that you can be serving video and stuff like that and that it will push it out to the edge servers that will be the most efficient. On the other hand, I’m not sure how that translates to VoIP and I’m surprised to learn that they’re able to do the real time. Who knew, you know. That’s the way I feel about it. So it’s kind of surprising that they have the capacity—you don’t have any problem with call quality apparently, right? Nir Simionovich: No. Well, let’s say this, as long as you’re working with carriers which were interconnected with, let’s say… [Music playing] Moderator: Yeah. Let me try to find the source of that. Nir Simionovich: OK. Moderator: That may have been DGM. Hello, DGM. Thank you. They may not. Sorry. Nir Simionovich: That was a good one. Moderator: Yeah. Well, that’s the Asterisk built-in… Nir Simionovich: We can talk over the music. Moderator: Frankly, I think that was somebody in Huntsville or else I just happened to click it by accident. Sorry. That happens from time to time. Nir Simionovich: Yeah. Moderator: Where were we? Nir Simionovich: In any case—where were we? Yeah, in any case, we were working with three or four different providers in the U.S. which are considered somewhere around, let’s say the tier 2. Then, usually, these carriers are interconnected on the backbone with links that are very, very efficient and reliable over to Amazon EC2 as long as you’re hosting your instances in the U.S. So, the end result that were, let’s say one of the carriers that we were using was called TSG Global and we were getting about 24 millisecond roundtrip from Amazon EC2 to TSG Global. Now, you’ll run voice over IP on that and will have perfect quality always. Moderator: I’m trying of course the agent to call in, trying to figure out because it would be more interesting to have a discussion here than reading the stuff in IRC. Nir Simionovich: Yeah, absolutely. Moderator: OK. My arguments have won over. He’s going to call. He or she, you never know, right? Nir Simionovich: Well, he brought up a very interesting point of how to bundle your own AMI and this is actually what I was talking about, creating your own AMI image. It’s not that simple. I have to be honest. It took me about four or five hours to actually understand how to do it right and it never works from the first try. It’s really annoying. What happens is that you need to take your running machine, your running AMI and you’re creating a raw06:25 image of the hard drive, of the virtual hard drive and then you upload that into S3. That’s complicated to do but if done right it will work on the second or third try. It never works on the first try, never. Moderator: OK and we have a bleep from Southwest British Columbia. Can you hear us? Could be the agent, you never know. Nir Simionovich: Actually, I’d love to hear about what he just wrote on Openfire XMPP and Red5 Flash Server for video conferencing. That is really interesting because I’ve been trying to play around with Red5 but I have to admit that Flash isn’t really my thing. So, I’d love to hear about that. Moderator: OK, the agent is coming out of the wild of Southern California. Who is in the Southwest British Columbia? Say hello. Come on, don’t be shy. We’re waiting for the agent to call in. We have time. Man 2: I’m here. Moderator: Yeah. Man 2: What do you do for your logs then here, just logs if you’re just bringing up these instances and using for so many hours and shutting them down? Are you uploading those at some point, back up to S3 or do you just not care about them? Nir Simionovich: Well, the application as is doesn’t really carry that much logging into the actual server itself but I have to admit that I never really needed to go about and offload the logs anywhere because we don’t really are about those. CDR’s are kept completely separated from the system but you could offload those into a different machine or you can, before shutting down the system, go back and offload those out of EC2. It shouldn’t be that much of a problem. Or if you really want to, you can use the—it’s not S3—what was that again—it’s called… Moderator: CloudFront? Nir Simionovich: No, not CloudFront. They’ve got servers where you can actually map a storage device to your AMI and then you can share&#8230;</p>
<p>Open Fire XMPP and the Red Five. Let&#8217;s get into that. You guys&#8230; Sure. &#8230;go ahead. I mean, I&#8217;d love to hear about that one. Yeah, so. Well, let me, let me, kind of, lay out the landscape a little bit for why I would even go down this path. I don&#8217;t know if you&#8217;ve heard of a specific podcast or video podcast that&#8217;s done and has been done for a couple of years, called Gilmor Gang It&#8217;s run by a guy named Steve Gilmore who&#8217;s a well know technology columnist here in, or up in Silicon Valley. Anyways, I&#8217;m basically an executive producer of that show and for the longest period of time we&#8217;ve been trying to put together, you know, a remote TV studio on the fly. And the biggest thing with this scenario without doing any hardware was really the concept of having, having remote individuals around the world that could dial in, in whatever manner they had available to them. If they had the ability to use a video teleconference capability either through a flash or through a Skype connection or if they just had a basic call-in functionality we needed to incorporate all those things because we wanted to see, we want to use a remote video teleconference capability as an actual remote camera so that we can bring them into the show. We can switch in and out of the different guests because generally Gilmor Gang handles 4-6 different guests on a given show. There&#8217;s a lot of banter. So, really, this has been an ongoing investigation of mine for probably over a year. And I&#8217;ve been using EC2&#8242;s primarily for, at this point, for Personis itself. That is completely hosted on EC2 all the way, all the way through. S3, EC2 SQS. I mean, I probably use almost every service floating around inside of Amazon at this point. And, I also had played a number of months ago, using, putting Asterisk up and running it there, playing around with it to see how well that would work. Which, it works extremely well. I mean, you&#8217;re able to take advantage of Amazon&#8217;s significant infrastructure, their tremendous pipe, that they have going throughout their data centers and their multiple peering relationships that they have for Internet access. So, I was very pleased with that. So we&#8217;re moving on to the next go-round. Okay. Hold on. Sorry about that. It&#8217;s okay. It&#8217;s hard to predict. But once I see the wave front, it&#8217;s over. Go ahead. Anyways. I thought, I thought the angels were coming for me or something. So. The a, so anyway, we&#8217;ve gone to this capability in an attempt to basically create this real-time audio-video capability that can be brought up or down on the fly. And that&#8217;s, kind of, the real story. Because, I mean, as he was mentioning regarding the phone, the service that they were running where it&#8217;s, they only need it for a short period of time. That&#8217;s the extreme value of Amazon. I mean, that&#8217;s where you really, really, really can take advantage of the system. Because you only need to bring up an instance for as long as you need it and then you can bring the instance down. You know, you can use it in such a, in that manner for a [R?] escape function, so that if you have, you know you&#8217;re going to have a burst of traffic coming in. You know, depending on how you set it up, you can pre-stage servers sitting there waiting for the burst or you can even do, you can get a little more sophisticated and create an auto-scale mechanism that, as traffic builds, it automatically instantiates new instances to handle the traffic burst and shuts them down accordingly as the traffic falls off. So, I mean, if you looking at, just to give you a price understanding, from an Amazon perspective. Just on a small instance. If you&#8217;re running an instance. The smallest instance that Amazon has. It costs you approximately, and this is without, you know, significant bandwidth going in or out. It&#8217;s only $70, it&#8217;s about 70 bucks a month. Now, the real significance, though, is that if you don&#8217;t need it up all the time, take it down. And that&#8217;s, kind of, what we&#8217;re doing with this whole remote video communication capability, in that, we can pull-up or pull-down anybody actually. We&#8217;re setting it up so that we can have remote. If somebody else wants to do a show that we want to add to our network, we&#8217;re going to have this all pre-configured so that they&#8217;ll have, they&#8217;ll have video-conferencing, they&#8217;ll have, and that&#8217;s the whole Red Five to XMPP integration. So we have our own chat functionality through XMPP. Don&#8217;t need IRC. We also have all the video capability that is, and SIP capabilities that are inherent inside of XMPP itself with Open Fire. And then we have the complete telecommunications capability that we&#8217;re embedding with Asterisk. And then, of course, the Red Five stuff, which is really handling the, you know, the multiple videos simultaneously function. That&#8217;s why the Red Five exists. So that I can have a room, literally, of ten, you know, video, teleconference, or video pieces coming in from all the participants. So those are the types of things that we&#8217;re currently working, or I&#8217;m currently on, that hopefully, we&#8217;ll see as, basically, a package that we&#8217;re going to load out into the world, that people can run. And that&#8217;s the other part of it. You can take advantage of the really big thing about the AMI&#8217;s, the other big part that I take a lot of, that I take advantage of all the time, is just that there&#8217;s a lot of people out there packaging these AMI&#8217;s. They, if they put them out in the public space, you can just go and click on one of them and say, &#8220;hey, what is this one, oh, look, there&#8217;s a Red Five server already configured by somebody. I&#8217;m going to go ahead and launch that AMI.&#8221; You can customize that AMI to your heart&#8217;s content and then ultimately save that to your, save that yourself as a private version of that instance. There&#8217;s a vast amount of power sitting within the Amazon infrastructure to take advantage of.</p>
<p>Nir: Jerry, one small question in regards to Red5, and this is something I have been toying around with, and maybe you’ll have an answer, because I didn’t get much work into it. Are you working with the, by any chance, with the Fontventa extensions for 3G, 3.5G cellular video transmission? Jerry: No I’m not. It’s an interesting… I’ve seen the extensions, I haven’t played with them at all. That would be kind of a next evolution for us. The other thing is that right now I’m kind of using, it’s kind of a limited version of Red5, in the sense that I’m actually running, and I didn’t even know this could exist until I found the ability through some board. There is actually, you can take Openfire and you can actually run Red5. Somebody’s written an extension that actually runs Red5 inside of Openfire. So, you don’t need to set up multiple servers. It’s basically an instance that’s prepackaged and you can basically bring it up and you’ve got full blown Jabber and you’ve got full blown Open5 sitting there, I mean Red5 sitting there, ready for you to use. The extensions that could take advantage of 3G, 3.5G, and 4, I mean those are all things that would be very, very cool to be able to put in. Because as you can see with things KiteTV and QIK, and a bunch of the other ones, you know there is no doubt that kind of capability from a video teleconference capability is going to be big, and big in the very near term.<br />
Nir: I have to admit that I didn’t have much time to play around with… Well l did play around with the Fontventa extensions, and I’ve done some work with that, in trying to integrate video, IP Video cams and Asterisk, and have, I’d say, some fair success with that. The thing is a lot of people now are talking about it. They are actually integrating a flash environment back into your cellular phone over 3G instead of writing your complete IBPR and doing that through Red5 seems like a good idea. But I will have to admit, that I hadn’t been playing with that lately. Again it’s very reassuring for me saying – OK there’s actually somebody out there really doing it. I mean, really progressing in that direction and we’re doing that on top of EZ2 is really exciting. Jerry: Yeah, and we think it’s really powerful, too. Again, the concept is that once this is a kind of an operational tapestry that we’ve kind of, or are orchestrating, that we really plan on bundling it as a kind of a preconfigured service that people in our… You know the concept, as we can all tell, media is changing for the worst, for the better, however you want to look at it doesn’t really matter to me, but bottom line is this entire users interface content and community that’s floating around out there and taking over, and watching mainstream media die. I mean, the idea that we can have this full blown, on the fly network capability brought up at will any where in the world and have multiple correspondence with basically multiple instances of both video and voice, there is an exorbitant amount of power there, especially when you start looking at what’s happening with the real time, in the real-time space, with things like Twitter and others. So, I think this whole new real-time space is actually going to usher in with Twitter and others, the ability to usher in this whole new paradigm for real-time, instantaneous, dynamic virtual, in essence radio stations, television stations, that can pop up on the fly and shut down on the fly at will. And it’s all going to be run through, basically, a cloud. So, that’s kind of what I’m focusing in on.<br />
Nir: Well I totally agree that there is a shift. Where we can all… well you know people who are already in it and are dealing with the shift, they are all feeling it pretty strongly. I remember when they started talking about cloud computing here in Israel that actually, the thing is that I was in this, there’s the Grid Community here in Israel, they were doing this event and I was looking at it saying – Well it’s pretty interesting but it doesn’t do much for real-time. Then I got back home and started to talk again with John Todd and saying – Well why not do it? It’s pretty interesting that there are real usage of real-time applications that can be done on the cloud and that’s really interesting to me, and really exciting. Not only from a commercial point of view or the economical point of view, but also from a technical point of view, because we have to rethink every thing. Where we’re really used to using like this enormous servers and saying – OK I’ve got 8 cores, I’ve got 16 cores, I’ve got 32 gigabytes of ram, and suddenly we don’t have that and it brings up a lot of the old paradigms that I used to work on back in the days of lets say Commodore 64 and saying – OK we need to be something really, really optimal, and the optimization is really interesting. It’s bringing back really old paradigms in terms of optimizations and really new paradigms in terms of economics.</p>
<p>Jerry: Yeah, yeah. I agree with you. I think what’s been happening, I mean it’s kind of how software leads hardware in some ways. You know, over the course of the past decade or more, the extreme focus on, or at least in the places I’ve played, a very laser guided focus on abstraction and decoupling. I mean that’s been a big part of a lot of systems that I’ve been a part of in the past 10 years. I think now it’s interesting to see cloud computing and the way it’s basically being pushed out from a hardware standpoint and how all the aspects of abstraction and decoupling are instrumental in being able to do these systems. Because you really do need a fairly significant abstractional error because you don’t know where the hardware exists. You don’t know how much memory you’re necessarily playing with. So, this whole decoupling is pretty paramount but it’s also been a trend that’s been going on for at least a decade or more.</p>
<p>Nir: Yeah, it’s been around here but not many people were using it, you know. I started my, let’s say, I started using the main frame computer somewhere around the age 18 or 19, when I was in the Army, and decoupling applications and the world of the main frame just sounds logical. It’s there, there’s nothing else. But when you are used to working on PC’s and servers, decoupling applications looked really weird. When you come to the company saying – OK you need to decouple this and they go – What? What are you talking about? What I don’t have access to the data base? That doesn’t sound right. The paradigm shift is going in a few directions and I’ve seen some companies that they, company like Giga Spaces, it’s an Israeli company, and they build like this enormous middle layer between your application and the Amazon EC2 cloud, which gives you the, let’s say, the imaginary effect of working inside a coupled system while you are completely decoupled. Which is really interesting. Jerry: Yeah, yeah.<br />
Nir: There is a lot of work being done right now. I think we are heading, in terms of technology, we’re heading for the next, I believe, is like the next three years is going to be really interesting in terms of cloud computing. Jerry: Yeah. I think the biggest thing for me, I’m somewhat of serial entrepreneur, and have been that way for a couple of decades now, but I think the thing that really excited me about the advent of things like EZ2 or Rackspace’s mossomo, or any of the other ones that are floating around there. There’s Slicehost which actually is now owned by Rackspace. It’s just the cost at which somebody could bootstrap up something, an idea, a concept. To be able to sit there and create a new business in this new paradigm and do it at a cost that is unheard of. Having done a few startups in my lifetime, you ultimately throw out an inurnment amount of money upfront just to handle, from a capital standpoint, the hardware that you are going to run on, especially if you’re a service offering. So I mean, that’s one of the things here, where you can get away with some, a couple of weeks worth of coding, you know, a $70 fee and you’ve got a service up and running in an hour or more, or a little bit more. But it’s nothing. It’s nothing to do and it’s cost effective, and if it doesn’t pan out, you basically shut down the AMI and you walk away from it.</p>
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			<wfw:commentRss>http://www.voipusersconference.org/2009/nir-simionovich-ec2/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-224823.mp3" length="29665462" type="audio/mpeg" />
			<itunes:keywords>Nir Simionovich,Amazon EC2</itunes:keywords>
		<itunes:subtitle>There is a full text transcript of this hour below. - Part 1: (Talkshoe)  Part 2 (ZipDX recording served from CloudFront):  - We&#039;ve tried to get people to talk about this before, but we didn&#039;t get a deep explanation. Here&#039;s a chance to ask Nir,</itunes:subtitle>
		<itunes:summary>There is a full text transcript of this hour below. (http://www.voipusersconference.org/2009/06/nir-simionovich-ec2/#more-665)

Part 1: (Talkshoe) 
Part 2 (ZipDX recording served from CloudFront): 

</itunes:summary>
		<itunes:author>randulo</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:00:00</itunes:duration>
	</item>
		<item>
		<title>Jim Van Meggelen on Building a solid system without the complexity</title>
		<link>http://www.voipusersconference.org/2009/jmv-build-system/</link>
		<comments>http://www.voipusersconference.org/2009/jmv-build-system/#comments</comments>
		<pubDate>Thu, 28 May 2009 18:00:41 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Intel Atom]]></category>
		<category><![CDATA[Jim Van Meggelen]]></category>
		<category><![CDATA[Mini ITX]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=575</guid>
		<description><![CDATA[Part 1: [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-224820.mp3] Part 2: [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-229501.mp3] Jim Van Meggelen is one of the author&#8217;s of O&#8217;Reilly&#8217;s Asterisk: The Future of Telephony, and an old school PBX guy. Jim is probably a bit of a masochist, which would explain why he got into the telecom business in the first place, and why he now loves Asterisk. [...]]]></description>
			<content:encoded><![CDATA[<p></p><p><a href="http://www.iconverged.com/"><img class="alignright size-full wp-image-589" title="Jim Van Meggelen" src="http://www.voipusersconference.org/wp-content/uploads/2009/05/jimvanm.jpg" alt="Jim Van Meggelen" width="163" height="256" /></a></p>
<p>Part 1:</p>
<p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-224820.mp3]</p>
<p>Part 2:</p>
<p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-229501.mp3]</p>
<p><a href="http://www.coretel.ca/jb/">Jim Van Meggelen</a> is one of the author&#8217;s of O&#8217;Reilly&#8217;s <a title="Second Edition!" href="http://oreilly.com/catalog/9780596510480/" target="_blank">Asterisk: The Future of Telephony</a>, and an old school PBX guy. Jim is probably a bit of a masochist, which would explain why he got into the telecom business in the first place, and why he now loves Asterisk. Jim is pretty friendly, kinda like a puppy that gets your shoes dirty. His enthusiasm is infectious, but also a little bit frightening if you stand too close. Jim is a partner in <a title="Core Telecom Innovations" href="http://www.coretel.ca/" target="_blank">Core Telecom Innovations</a>, a recognized leader in the Asterisk Telephony Revolution, and <a title="iConverged, Inc" href="http://www.iconverged.com/" target="_blank">iConverged Inc</a>. He lives in Toronto with his wife and three kids, and loves writing, photography, speaking, improv, choral singing, and old shoes.</p>
<p>Jim will talk about building an embedded-like system using x86 hardware (Intel Atom Mini-ITX) and a stock Linux distro for well under $1000, including FX cards with hardware echo cancellation &#8212; this is a great way to build a system that&#8217;s almost like an embedded system, but with much less software complexity or incompatibility.</p>
<p><a href="http://www.voipusersconference.org/wp-content/uploads/2009/05/irc2009-05-29.txt">IRC TRanscript 2009-05-29</a></p>
]]></content:encoded>
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		<slash:comments>0</slash:comments>
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			<itunes:keywords>asterisk,Intel Atom,Jim Van Meggelen,Mini ITX</itunes:keywords>
		<itunes:subtitle>Part 1: - Part 2: - Jim Van Meggelen is one of the author&#039;s of O&#039;Reilly&#039;s Asterisk: The Future of Telephony, and an old school PBX guy. Jim is probably a bit of a masochist, which would explain why he got into the telecom business in the firs...</itunes:subtitle>
		<itunes:summary>(http://www.voipusersconference.org/wp-content/uploads/2009/05/jimvanm.jpg)

Part 1:



Part 2:



Jim Van Meggelen (http://www.coretel.ca/jb/) is one of the author&#039;s of O&#039;Reilly&#039;s Asterisk: The Future of Telephony (http://oreilly.com/catalog/9780596510480/), and an old school PBX guy. Jim is probably a bit of a masochist, which would explain why he got into the telecom business in the first place, and why he now loves Asterisk. Jim is pretty friendly, kinda like a puppy that gets your shoes dirty. His enthusiasm is infectious, but also a little bit frightening if you stand too close. Jim is a partner in Core Telecom Innovations (http://www.coretel.ca/), a recognized leader in the Asterisk Telephony Revolution, and iConverged Inc (http://www.iconverged.com/). He lives in Toronto with his wife and three kids, and loves writing, photography, speaking, improv, choral singing, and old shoes.

Jim will talk about building an embedded-like system using x86 hardware (Intel Atom Mini-ITX) and a stock Linux distro for well under $1000, including FX cards with hardware echo cancellation -- this is a great way to build a system that&#039;s almost like an embedded system, but with much less software complexity or incompatibility.

IRC TRanscript 2009-05-29 (http://www.voipusersconference.org/wp-content/uploads/2009/05/irc2009-05-29.txt)</itunes:summary>
		<itunes:author>randulo</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
		<itunes:duration>1:00:00</itunes:duration>
	</item>
		<item>
		<title>Jason Goecke, Adhearsion</title>
		<link>http://www.voipusersconference.org/2009/jason-adhearsion/</link>
		<comments>http://www.voipusersconference.org/2009/jason-adhearsion/#comments</comments>
		<pubDate>Mon, 04 May 2009 14:40:36 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[video]]></category>
		<category><![CDATA[VoIP]]></category>
		<category><![CDATA[Adhearsion]]></category>
		<category><![CDATA[applications]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Jason Goecke]]></category>
		<category><![CDATA[Jay Phillips]]></category>
		<category><![CDATA[Ruby]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=518</guid>
		<description><![CDATA[Here&#8217;s a chance to meet Jason but unfortunately, you can&#8217;t hear the jokes and stories we all shared at dinner. Adhearsion is another one of those forward-looking companies that shakes up the VoIP world and adds spice and inspiration to a developer&#8217;s life. Adhearsion is a new way to write voice-enabled applications. It&#8217;s not just [...]]]></description>
			<content:encoded><![CDATA[<p></p><p>Here&#8217;s a chance to meet Jason but unfortunately, you can&#8217;t hear the jokes and stories we all shared at dinner. <a href="http://www.adhearsion.com">Adhearsion</a> is another one of those forward-looking companies that shakes up the VoIP world and adds spice and inspiration to a developer&#8217;s life.</p>
<p><img src="http://www.voipusersconference.org/wp-content/plugins/flash-video-player/default_video_player.gif" /></p>
<div class="wp-caption alignright" style="width: 240px">
	<a href="http://www.flickr.com/photos/14858094@N00/385881625"><img style="border: 0pt none;" title="Jay Philips from Adhearsion" src="http://farm1.static.flickr.com/154/385881625_e6354e85e1_m.jpg" border="0" alt="Me in Linux Journal Magazine" hspace="5" width="240" height="180" /></a>
	<p class="wp-caption-text">The other Adhearsion guy,  Jay Philips</p>
</div>
<p>Adhearsion is a new way to write voice-enabled applications. It&#8217;s not just an API or library — it&#8217;s a fully-featured framework, the first of its kind, designed for maximal code reuse and intuitiveness. The name &#8220;Adhearsion&#8221; is a combination of &#8220;adhesion&#8221; and &#8220;hear&#8221; because Adhearsion shines best when integrating technologies with voice.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2009/jason-adhearsion/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Skype for Asterisk with Steve Sokol</title>
		<link>http://www.voipusersconference.org/2008/skype-for-asterisk/</link>
		<comments>http://www.voipusersconference.org/2008/skype-for-asterisk/#comments</comments>
		<pubDate>Fri, 26 Dec 2008 12:00:31 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[digium]]></category>
		<category><![CDATA[skype]]></category>

		<guid isPermaLink="false">http://sessions.voipusersconference.org/?p=210</guid>
		<description><![CDATA[Image by alexmuse via Flickr [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-170693.mp3] Skype for Asterisk with Steve Sokol, Digium product manager. Also a general talk about life, the universe and everything such as the high points of VoIP in 2008. Related articles by Zemanta Digium Offers Subscription Support Plans for Asterisk (ostatic.com) Skype for SIP == Skype for Asterisk DOA? (skypejournal.com) [...]]]></description>
			<content:encoded><![CDATA[<p></p><div class="zemanta-img" style="margin: 1em; display: block;">
<div>
<dl class="wp-caption alignright" style="width: 250px;">
<dt class="wp-caption-dt"><a href="http://www.flickr.com/photos/93504305@N00/279414177"><img title="Digium" src="http://farm1.static.flickr.com/97/279414177_02f779df60_m.jpg" alt="Digium" width="240" height="160" /></a></dt>
<dd class="wp-caption-dd zemanta-img-attribution" style="font-size: 0.8em;">Image by <a href="http://www.flickr.com/photos/93504305@N00/279414177">alexmuse</a> via Flickr</dd>
</dl>
</div>
</div>
<p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-170693.mp3]</p>
<p><span id="EpisodePopEpisodeDescriptionLabel" class="LabelText"> </span><span id="EpisodePopEpisodeDescription">Skype for Asterisk with Steve Sokol, Digium product manager. Also a general talk about life, the universe and everything such as the high points of VoIP in 2008.</span></p>
<h6 class="zemanta-related-title" style="font-size: 1em;">Related articles by Zemanta</h6>
<ul class="zemanta-article-ul">
<li class="zemanta-article-ul-li"><a href="http://ostatic.com/blog/digium-offers-subscription-support-plans-for-asterisk">Digium Offers Subscription Support Plans for Asterisk</a> (ostatic.com)</li>
<li class="zemanta-article-ul-li"><a href="http://skypejournal.com/2009/03/skype-for-sip-skype-for-asterisk-doa.html">Skype for SIP == Skype for Asterisk DOA?</a> (skypejournal.com)</li>
<li class="zemanta-article-ul-li"><a href="http://www.lucafiligheddu.com/2009/03/top-30-voip-leaders-on-twitter.html">Top 30 VoIP Leaders On Twitter</a> (lucafiligheddu.com)</li>
</ul>
<div class="zemanta-pixie" style="margin-top: 10px; height: 15px;"><a class="zemanta-pixie-a" title="Reblog this post [with Zemanta]" href="http://reblog.zemanta.com/zemified/d8d09582-548e-4c8f-84af-16b412194d69/"><img class="zemanta-pixie-img" style="border: medium none; float: right;" src="http://img.zemanta.com/reblog_e.png?x-id=d8d09582-548e-4c8f-84af-16b412194d69" alt="Reblog this post [with Zemanta]" /></a><span class="zem-script more-related pretty-attribution"><script src="http://static.zemanta.com/readside/loader.js" type="text/javascript"></script></span></div>
]]></content:encoded>
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		<slash:comments>0</slash:comments>
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			<itunes:keywords>asterisk,digium,skype</itunes:keywords>
		<itunes:subtitle>Image by alexmuse via Flickr  Skype for Asterisk with Steve Sokol, Digium product manager. Also a general talk about life, the universe and everything such as the high points of VoIP in 2008. Related articles by Zemanta - </itunes:subtitle>
		<itunes:summary>(http://farm1.static.flickr.com/97/279414177_02f779df60_m.jpg) Image by alexmuse (http://www.flickr.com/photos/93504305@N00/279414177) via Flickr 



 Skype for Asterisk with Steve Sokol, Digium product manager. Also a general talk about life, the universe and everything such as the high points of VoIP in 2008.
Related articles by Zemanta

	Digium Offers Subscription Support Plans for Asterisk (http://ostatic.com/blog/digium-offers-subscription-support-plans-for-asterisk) (ostatic.com)
	Skype for SIP == Skype for Asterisk DOA? (http://skypejournal.com/2009/03/skype-for-sip-skype-for-asterisk-doa.html) (skypejournal.com)
	Top 30 VoIP Leaders On Twitter (http://www.lucafiligheddu.com/2009/03/top-30-voip-leaders-on-twitter.html) (lucafiligheddu.com)

(http://img.zemanta.com/reblog_e.png?x-id=d8d09582-548e-4c8f-84af-16b412194d69)</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>Sources of Asterisk Knowledge</title>
		<link>http://www.voipusersconference.org/2008/sources-of-asterisk-knowledge/</link>
		<comments>http://www.voipusersconference.org/2008/sources-of-asterisk-knowledge/#comments</comments>
		<pubDate>Fri, 03 Oct 2008 12:00:21 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Books]]></category>
		<category><![CDATA[chats]]></category>
		<category><![CDATA[documentation]]></category>
		<category><![CDATA[forums]]></category>
		<category><![CDATA[podcasts]]></category>

		<guid isPermaLink="false">http://sessions.voipusersconference.org/?p=263</guid>
		<description><![CDATA[[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-146979.mp3] How can asterisk users get up to speed? Let us count the ways. The books, chats, podcasts, mailing lists, forums and other resources available to make it easier for you to master Asterisk and VoIP. Foremost link: Voip-Info.org]]></description>
			<content:encoded><![CDATA[<p></p><p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-146979.mp3]</p>
<p>How can asterisk users get up to speed? Let us count the ways.</p>
<p>The books, chats, podcasts, mailing lists, forums and other resources available to make it easier for you to master Asterisk and VoIP.</p>
<p>Foremost link: <a href="http://voip-info.org">Voip-Info.org</a></p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2008/sources-of-asterisk-knowledge/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
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			<itunes:keywords>asterisk,Books,chats,documentation,forums,podcasts</itunes:keywords>
		<itunes:subtitle>How can asterisk users get up to speed? Let us count the ways. - The books, chats, podcasts, mailing lists, forums and other resources available to make it easier for you to master Asterisk and VoIP. - Foremost link: Voip-Info.org</itunes:subtitle>
		<itunes:summary>How can asterisk users get up to speed? Let us count the ways.

The books, chats, podcasts, mailing lists, forums and other resources available to make it easier for you to master Asterisk and VoIP.

Foremost link: Voip-Info.org (http://voip-info.org)</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>Asterisk Scaling with Clusters</title>
		<link>http://www.voipusersconference.org/2008/asterisk-scaling-with-clusters/</link>
		<comments>http://www.voipusersconference.org/2008/asterisk-scaling-with-clusters/#comments</comments>
		<pubDate>Fri, 27 Jun 2008 12:00:14 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[clusters]]></category>
		<category><![CDATA[installation]]></category>
		<category><![CDATA[large]]></category>
		<category><![CDATA[large installations]]></category>
		<category><![CDATA[scaling]]></category>

		<guid isPermaLink="false">http://sessions.voipusersconference.org/?p=247</guid>
		<description><![CDATA[[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-126958.mp3] J.R. Richardson joins us for an extended talk about using multiple instances of asterisk for scaling. Jeremy from Nufone also had some great input on virtualization.]]></description>
			<content:encoded><![CDATA[<p></p><p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-126958.mp3]</p>
<p> J.R. Richardson joins us for an extended talk about using multiple instances of asterisk for scaling. Jeremy from Nufone also had some great input on virtualization.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2008/asterisk-scaling-with-clusters/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-126958.mp3" length="34653073" type="audio/mpeg" />
			<itunes:keywords>asterisk,clusters,installation,large,large installations,scaling</itunes:keywords>
		<itunes:subtitle>J.R. Richardson joins us for an extended talk about using multiple instances of asterisk for scaling. Jeremy from Nufone also had some great input on virtualization.</itunes:subtitle>
		<itunes:summary>J.R. Richardson joins us for an extended talk about using multiple instances of asterisk for scaling. Jeremy from Nufone also had some great input on virtualization.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>Asterisk Large Installations with Mike Trest</title>
		<link>http://www.voipusersconference.org/2008/asterisk-large-installations-with-mike-trest/</link>
		<comments>http://www.voipusersconference.org/2008/asterisk-large-installations-with-mike-trest/#comments</comments>
		<pubDate>Fri, 23 May 2008 12:00:09 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[large installations]]></category>
		<category><![CDATA[Mike Tresk]]></category>

		<guid isPermaLink="false">http://sessions.voipusersconference.org/?p=229</guid>
		<description><![CDATA[[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-117256.mp3] After many requests, we finally have someone to talk on large scale implementation of VoIP systems with asterisk. Using a farm of Asterisk and Digium cards, tens of thousands of simultaneous calls can be made and Mike Trest has offered to take it all apart for us to look inside.]]></description>
			<content:encoded><![CDATA[<p></p><p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-117256.mp3]</p>
<p>After many requests, we finally have someone to talk on large scale implementation of VoIP systems with asterisk. Using a farm of Asterisk and Digium cards, tens of thousands of simultaneous calls can be made and <a href="http://mike.trest.com/">Mike Trest</a> has offered to take it all apart for us to look inside.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2008/asterisk-large-installations-with-mike-trest/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
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			<itunes:keywords>asterisk,large installations,Mike Tresk</itunes:keywords>
		<itunes:subtitle>After many requests, we finally have someone to talk on large scale implementation of VoIP systems with asterisk. Using a farm of Asterisk and Digium cards, tens of thousands of simultaneous calls can be made and Mike Trest has offered to take it all a...</itunes:subtitle>
		<itunes:summary>After many requests, we finally have someone to talk on large scale implementation of VoIP systems with asterisk. Using a farm of Asterisk and Digium cards, tens of thousands of simultaneous calls can be made and Mike Trest (http://mike.trest.com/) has offered to take it all apart for us to look inside.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>Asterisk 3rd Party Licensing</title>
		<link>http://www.voipusersconference.org/2008/asterisk-3rd-party-licensing/</link>
		<comments>http://www.voipusersconference.org/2008/asterisk-3rd-party-licensing/#comments</comments>
		<pubDate>Fri, 09 May 2008 12:00:06 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[3rd party licensing]]></category>
		<category><![CDATA[asterisk]]></category>

		<guid isPermaLink="false">http://sessions.voipusersconference.org/?p=251</guid>
		<description><![CDATA[[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-109845.mp3] The purpose of this call is to discuss the community&#8217;s feelings about an Asterisk 3rd party developed commercial software sales licensing platform. The plan is that some form of documented published schema be implemented that will allow for 3rd party software developers to sell their software applications using a common licensing model similar to [...]]]></description>
			<content:encoded><![CDATA[<p></p><p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-109845.mp3]</p>
<p>The purpose of this call is to discuss the community&#8217;s feelings about an Asterisk 3rd party developed commercial software sales licensing platform. The plan is that some form of documented published schema be implemented that will allow for 3rd party software developers to sell their software applications using a common licensing model similar to the way G729 licenses are sold by Digium.</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2008/asterisk-3rd-party-licensing/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
<enclosure url="http://media.blubrry.com/voipusers/media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-109845.mp3" length="32458057" type="audio/mpeg" />
			<itunes:keywords>3rd party licensing,asterisk</itunes:keywords>
		<itunes:subtitle>The purpose of this call is to discuss the community&#039;s feelings about an Asterisk 3rd party developed commercial software sales licensing platform. The plan is that some form of documented published schema be implemented that will allow for 3rd party s...</itunes:subtitle>
		<itunes:summary>The purpose of this call is to discuss the community&#039;s feelings about an Asterisk 3rd party developed commercial software sales licensing platform. The plan is that some form of documented published schema be implemented that will allow for 3rd party software developers to sell their software applications using a common licensing model similar to the way G729 licenses are sold by Digium.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>Asterisk and Fax</title>
		<link>http://www.voipusersconference.org/2008/asterisk-and-fax/</link>
		<comments>http://www.voipusersconference.org/2008/asterisk-and-fax/#comments</comments>
		<pubDate>Fri, 11 Apr 2008 12:00:35 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[fax]]></category>

		<guid isPermaLink="false">http://sessions.voipusersconference.org/?p=253</guid>
		<description><![CDATA[[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-105284.mp3] Part 1 Faxing with Asterisk [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-107305.mp3] Part 2]]></description>
			<content:encoded><![CDATA[<p></p><p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-105284.mp3] Part 1</p>
<p>Faxing with Asterisk</p>
<p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-107305.mp3] Part 2</p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2008/asterisk-and-fax/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
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			<itunes:keywords>asterisk,fax</itunes:keywords>
		<itunes:subtitle>Part 1 - Faxing with Asterisk Part 2</itunes:subtitle>
		<itunes:summary>Part 1

Faxing with Asterisk

 Part 2</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>Mark Spencer, Creator of Asterisk</title>
		<link>http://www.voipusersconference.org/2008/mark-spencer-creator-of-asterisk/</link>
		<comments>http://www.voipusersconference.org/2008/mark-spencer-creator-of-asterisk/#comments</comments>
		<pubDate>Fri, 04 Jan 2008 12:00:20 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[digium]]></category>
		<category><![CDATA[Mark Spencer]]></category>

		<guid isPermaLink="false">http://sessions.voipusersconference.org/?p=275</guid>
		<description><![CDATA[Image by jimgris via Flickr [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-78802.mp3] Does Mark need an introduction? I think not. He&#8217;ll be live to answer questions and talk about what&#8217;s going on with all things asterisk (which is a registered trademark of Digium) Related articles by Zemanta The future of open source (infoworld.com) Top 30 VoIP Leaders On Twitter (lucafiligheddu.com)]]></description>
			<content:encoded><![CDATA[<p></p><div class="zemanta-img" style="margin: 1em; display: block;">
<div>
<dl class="wp-caption alignright" style="width: 250px;">
<dt class="wp-caption-dt"><a href="http://www.flickr.com/photos/80887578@N00/100344793"><img title="OSBC San Francisco 06" src="http://farm1.static.flickr.com/29/100344793_c2c766fbeb_m.jpg" alt="OSBC San Francisco 06" width="240" height="160" /></a></dt>
<dd class="wp-caption-dd zemanta-img-attribution" style="font-size: 0.8em;">Image by <a href="http://www.flickr.com/photos/80887578@N00/100344793">jimgris</a> via Flickr</dd>
</dl>
</div>
</div>
<p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-78802.mp3]<br />
Does Mark need an introduction? I think not. He&#8217;ll be live to answer questions and talk about what&#8217;s going on with all things asterisk (which is a registered trademark of Digium)</p>
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]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2008/mark-spencer-creator-of-asterisk/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
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			<itunes:keywords>asterisk,digium,Mark Spencer</itunes:keywords>
		<itunes:subtitle>Image by jimgris via Flickr Does Mark need an introduction? I think not. He&#039;ll be live to answer questions and talk about what&#039;s going on with all things asterisk (which is a registered trademark of Digium) Related articles by Zemanta - </itunes:subtitle>
		<itunes:summary>(http://farm1.static.flickr.com/29/100344793_c2c766fbeb_m.jpg)Image by jimgris (http://www.flickr.com/photos/80887578@N00/100344793) via Flickr


Does Mark need an introduction? I think not. He&#039;ll be live to answer questions and talk about what&#039;s going on with all things asterisk (which is a registered trademark of Digium)
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(http://img.zemanta.com/reblog_e.png?x-id=c8c5286e-4860-4f86-8a37-68e338449365)</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>New Asterisk Book by Stefan Wintermeyer</title>
		<link>http://www.voipusersconference.org/2007/asterisk-book-stefan-wintermeyer/</link>
		<comments>http://www.voipusersconference.org/2007/asterisk-book-stefan-wintermeyer/#comments</comments>
		<pubDate>Fri, 08 Jun 2007 12:00:39 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[AMOOCON]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Asterisk Tag]]></category>
		<category><![CDATA[book]]></category>
		<category><![CDATA[Stefan Wintermeyer]]></category>

		<guid isPermaLink="false">http://sessions.voipusersconference.org/?p=293</guid>
		<description><![CDATA[[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-23423.mp3] Stefan and his translator Stephen Bosch will be with us to talk about this new book, already a best-seller in German and soon to appear in English. Stefan produced Asterisk Tag in 2008 now called AMOOCON in 2009.]]></description>
			<content:encoded><![CDATA[<p></p><p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-23423.mp3]</p>
<div class="wp-caption alignleft" style="width: 185px">
	<a href="http://amoocon.de/"><img title="Stefan Book" src="http://images.barnesandnoble.com/images/18510000/18516442.JPG" alt="Practical Asterisk 1.4 &amp; 1.6: From Beginner to Expert" width="185" height="245" /></a>
	<p class="wp-caption-text">Practical Asterisk 1.4 &amp; 1.6: From Beginner to Expert</p>
</div>
<p>Stefan and his translator Stephen Bosch will be with us to talk about this new book, already a best-seller in German and soon to appear in English. Stefan produced Asterisk Tag in 2008 now called <a href="http://www.amoocon.de">AMOOCON in 2009</a>.</p>
]]></content:encoded>
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		<slash:comments>0</slash:comments>
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			<itunes:keywords>AMOOCON,asterisk,Asterisk Tag,book,Stefan Wintermeyer</itunes:keywords>
		<itunes:subtitle>Stefan and his translator Stephen Bosch will be with us to talk about this new book, already a best-seller in German and soon to appear in English. Stefan produced Asterisk Tag in 2008 now called AMOOCON in 2009.</itunes:subtitle>
		<itunes:summary>Stefan and his translator Stephen Bosch will be with us to talk about this new book, already a best-seller in German and soon to appear in English. Stefan produced Asterisk Tag in 2008 now called AMOOCON in 2009 (http://www.amoocon.de).</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>Mark Spencer on the Appliance</title>
		<link>http://www.voipusersconference.org/2007/mark-spencer-on-the-appliance/</link>
		<comments>http://www.voipusersconference.org/2007/mark-spencer-on-the-appliance/#comments</comments>
		<pubDate>Thu, 19 Apr 2007 12:00:02 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[appliance]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[Mark Spencer]]></category>

		<guid isPermaLink="false">http://sessions.voipusersconference.org/?p=285</guid>
		<description><![CDATA[[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-17341.mp3] Mark talks about the asterisk appliance, asterisk in Japanese, development efforts and a lot of other interesting tidbits.]]></description>
			<content:encoded><![CDATA[<p></p><p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-17341.mp3]</p>
<p>Mark talks about the asterisk appliance, asterisk in Japanese, development efforts and a lot of other interesting tidbits. </p>
]]></content:encoded>
			<wfw:commentRss>http://www.voipusersconference.org/2007/mark-spencer-on-the-appliance/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
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			<itunes:keywords>appliance,asterisk,Mark Spencer</itunes:keywords>
		<itunes:subtitle>Mark talks about the asterisk appliance, asterisk in Japanese, development efforts and a lot of other interesting tidbits.</itunes:subtitle>
		<itunes:summary>Mark talks about the asterisk appliance, asterisk in Japanese, development efforts and a lot of other interesting tidbits.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
	</item>
		<item>
		<title>featuremap.conf, vonage lawsuit</title>
		<link>http://www.voipusersconference.org/2007/featuremap/</link>
		<comments>http://www.voipusersconference.org/2007/featuremap/#comments</comments>
		<pubDate>Thu, 12 Apr 2007 23:00:24 +0000</pubDate>
		<dc:creator>VUC</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[featuremap]]></category>
		<category><![CDATA[vonage]]></category>

		<guid isPermaLink="false">http://www.voipusersconference.org/?p=1669</guid>
		<description><![CDATA[[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-14228.mp3] featurmap and applicationmap, their uses. Ok, we were all over the map this time but some interesting stuff about iotum, CRM, encryption, verizon/vonage suit, etc.]]></description>
			<content:encoded><![CDATA[<p></p><p>[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-14228.mp3]<br />
featurmap and applicationmap, their uses. Ok, we were all over the map this time but some interesting stuff about iotum, CRM, encryption, verizon/vonage suit, etc.</p>
]]></content:encoded>
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		<slash:comments>0</slash:comments>
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			<itunes:keywords>asterisk,featuremap,vonage</itunes:keywords>
		<itunes:subtitle>featurmap and applicationmap, their uses. Ok, we were all over the map this time but some interesting stuff about iotum, CRM, encryption, verizon/vonage suit, etc.</itunes:subtitle>
		<itunes:summary>featurmap and applicationmap, their uses. Ok, we were all over the map this time but some interesting stuff about iotum, CRM, encryption, verizon/vonage suit, etc.</itunes:summary>
		<itunes:author>@voipusers</itunes:author>
		<itunes:explicit>clean</itunes:explicit>
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