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	<title>Comments on: Asterisk and FreeSWITCH</title>
	<atom:link href="http://www.os-voip.com/2008/08/asterisk-and-freeswitch/feed/" rel="self" type="application/rss+xml" />
	<link>http://www.os-voip.com/2008/08/asterisk-and-freeswitch/</link>
	<description>Open Source VoIP by Aaron Rosenthal</description>
	<lastBuildDate>Sat, 21 Aug 2010 18:11:29 +0000</lastBuildDate>
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		<title>By: Bart</title>
		<link>http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-755</link>
		<dc:creator>Bart</dc:creator>
		<pubDate>Thu, 14 Jan 2010 13:30:44 +0000</pubDate>
		<guid isPermaLink="false">http://www.os-voip.com/?p=101#comment-755</guid>
		<description>I agree with gitguy, don&#039;t use asterisk if your deployment is sip based. FS does much better on SIP transparency and RTP handling. I had a lot of issues with asterisk which are now resolved now on FS. Congrats to the FS development team.</description>
		<content:encoded><![CDATA[<p>I agree with gitguy, don&#8217;t use asterisk if your deployment is sip based. FS does much better on SIP transparency and RTP handling. I had a lot of issues with asterisk which are now resolved now on FS. Congrats to the FS development team.</p>
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		<title>By: Mark J Crane</title>
		<link>http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-754</link>
		<dc:creator>Mark J Crane</dc:creator>
		<pubDate>Thu, 19 Nov 2009 07:13:03 +0000</pubDate>
		<guid isPermaLink="false">http://www.os-voip.com/?p=101#comment-754</guid>
		<description>Like many others I started with Asterisk then learned FreeSWITCH. My favorite things about FreeSWITCH are that it can run natively on multiple platforms, its scallable, modular and very customizable. Its like a voip tool box with many pieces that can be put together to build whatever voip application you want. 

I like FreeSWITCH so much that I built an open source graphical interface for FreeSWITCH called FusionPBX.</description>
		<content:encoded><![CDATA[<p>Like many others I started with Asterisk then learned FreeSWITCH. My favorite things about FreeSWITCH are that it can run natively on multiple platforms, its scallable, modular and very customizable. Its like a voip tool box with many pieces that can be put together to build whatever voip application you want. </p>
<p>I like FreeSWITCH so much that I built an open source graphical interface for FreeSWITCH called FusionPBX.</p>
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		<title>By: TimCox</title>
		<link>http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-723</link>
		<dc:creator>TimCox</dc:creator>
		<pubDate>Sun, 04 Oct 2009 12:48:54 +0000</pubDate>
		<guid isPermaLink="false">http://www.os-voip.com/?p=101#comment-723</guid>
		<description>Also, FreeSwitch is the basis of CudaTel by Barracuda, I have Trixbox now w/Polycoms and am looking forward to the switchover/Integration at my office (Asterisk deadlocks SUCK).</description>
		<content:encoded><![CDATA[<p>Also, FreeSwitch is the basis of CudaTel by Barracuda, I have Trixbox now w/Polycoms and am looking forward to the switchover/Integration at my office (Asterisk deadlocks SUCK).</p>
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		<title>By: Open Source VoIP in the carrier space : A look at Bandwidth.com &#124; OS-VoIP</title>
		<link>http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-507</link>
		<dc:creator>Open Source VoIP in the carrier space : A look at Bandwidth.com &#124; OS-VoIP</dc:creator>
		<pubDate>Fri, 31 Oct 2008 21:58:55 +0000</pubDate>
		<guid isPermaLink="false">http://www.os-voip.com/?p=101#comment-507</guid>
		<description>[...] have nearly the same breadth of capabilities and support found in Asterisk. Take a gander at a comparison I wrote about the two. FreeSWITCH is what sits behind Bandwidth.com&#8217;s new PhoneBooth product, [...]</description>
		<content:encoded><![CDATA[<p>[...] have nearly the same breadth of capabilities and support found in Asterisk. Take a gander at a comparison I wrote about the two. FreeSWITCH is what sits behind Bandwidth.com&#8217;s new PhoneBooth product, [...]</p>
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	<item>
		<title>By: elibiordefibia</title>
		<link>http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-419</link>
		<dc:creator>elibiordefibia</dc:creator>
		<pubDate>Sun, 28 Sep 2008 11:32:54 +0000</pubDate>
		<guid isPermaLink="false">http://www.os-voip.com/?p=101#comment-419</guid>
		<description>favorited this one, dude</description>
		<content:encoded><![CDATA[<p>favorited this one, dude</p>
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	<item>
		<title>By: coloquic</title>
		<link>http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-400</link>
		<dc:creator>coloquic</dc:creator>
		<pubDate>Mon, 22 Sep 2008 18:32:18 +0000</pubDate>
		<guid isPermaLink="false">http://www.os-voip.com/?p=101#comment-400</guid>
		<description>favorited this one, brother</description>
		<content:encoded><![CDATA[<p>favorited this one, brother</p>
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		<title>By: H323 Ip Phone</title>
		<link>http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-382</link>
		<dc:creator>H323 Ip Phone</dc:creator>
		<pubDate>Sat, 20 Sep 2008 09:15:44 +0000</pubDate>
		<guid isPermaLink="false">http://www.os-voip.com/?p=101#comment-382</guid>
		<description>&lt;strong&gt;H323 Ip Phone...&lt;/strong&gt;

Very nice, I like your post....</description>
		<content:encoded><![CDATA[<p><strong>H323 Ip Phone&#8230;</strong></p>
<p>Very nice, I like your post&#8230;.</p>
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		<title>By: Anthony Minessale</title>
		<link>http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-52</link>
		<dc:creator>Anthony Minessale</dc:creator>
		<pubDate>Mon, 11 Aug 2008 20:09:58 +0000</pubDate>
		<guid isPermaLink="false">http://www.os-voip.com/?p=101#comment-52</guid>
		<description>Being the Author of both FreeSWITCH and a long list of Asterisk 1.2 features I can offer up my opinion on the matter for what it&#039;s worth.

I was asked so many times why I bothered with FreeSWITCH that I wrote the following article to explain it:
http://www.freeswitch.org/node/117

Being an Asterisk developer myself who still hosts a bunch of my old asterisk apps, ( http://www.freeswitch.org/node/50 )
my only concern with all this is the perception of some sort of Open Source &quot;turf war&quot; about what software is &quot;better&quot; etc.  When I decide what open source software I use I judge it purely on if it&#039;s the *right tool for the job* (tm).  I decided Asterisk was not the right tool only after 3 years of using it an learning it inside and out hence why I started FreeSWITCH.  Many may find Asterisk more than adequate or maybe they use Asterisk, FreeSWITCH and OpenSER all together in an enterprise system.  Why should we care? We are all on the same team..The Open Source VoIP team.</description>
		<content:encoded><![CDATA[<p>Being the Author of both FreeSWITCH and a long list of Asterisk 1.2 features I can offer up my opinion on the matter for what it&#8217;s worth.</p>
<p>I was asked so many times why I bothered with FreeSWITCH that I wrote the following article to explain it:<br />
<a href="http://www.freeswitch.org/node/117" rel="nofollow">http://www.freeswitch.org/node/117</a></p>
<p>Being an Asterisk developer myself who still hosts a bunch of my old asterisk apps, ( <a href="http://www.freeswitch.org/node/50" rel="nofollow">http://www.freeswitch.org/node/50</a> )<br />
my only concern with all this is the perception of some sort of Open Source &#8220;turf war&#8221; about what software is &#8220;better&#8221; etc.  When I decide what open source software I use I judge it purely on if it&#8217;s the *right tool for the job* &#8482;.  I decided Asterisk was not the right tool only after 3 years of using it an learning it inside and out hence why I started FreeSWITCH.  Many may find Asterisk more than adequate or maybe they use Asterisk, FreeSWITCH and OpenSER all together in an enterprise system.  Why should we care? We are all on the same team..The Open Source VoIP team.</p>
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	<item>
		<title>By: asdx</title>
		<link>http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-50</link>
		<dc:creator>asdx</dc:creator>
		<pubDate>Sun, 10 Aug 2008 22:16:42 +0000</pubDate>
		<guid isPermaLink="false">http://www.os-voip.com/?p=101#comment-50</guid>
		<description>Some things I would like to have in FS:

1- an easy way to dial out from conferences, from the phone-side.

2- the ability to set per-bridge/leg timeout, along with global timeout.

3- complete IAX/H.323 stack that supports registration, trunking and everything.

4- &quot;sip/iax2 show peer &quot; to see user settings from the cli.</description>
		<content:encoded><![CDATA[<p>Some things I would like to have in FS:</p>
<p>1- an easy way to dial out from conferences, from the phone-side.</p>
<p>2- the ability to set per-bridge/leg timeout, along with global timeout.</p>
<p>3- complete IAX/H.323 stack that supports registration, trunking and everything.</p>
<p>4- &#8220;sip/iax2 show peer &#8221; to see user settings from the cli.</p>
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	<item>
		<title>By: gitguy</title>
		<link>http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-49</link>
		<dc:creator>gitguy</dc:creator>
		<pubDate>Sun, 10 Aug 2008 20:38:48 +0000</pubDate>
		<guid isPermaLink="false">http://www.os-voip.com/?p=101#comment-49</guid>
		<description>You are wrong.  Asterisk sucks in many ways, it has flaws, I&#039;ve tried to deploy it and found a million of bugs, which are already reported in mantis.

Not to mention that the SIP implementation in Asterisk sucks (chan_sip). whereas Sofia-SIP in FreeSWITCH is a lot better, and your comment about FreeSWITCH lacking in features is FUD.

Asterisk sucks, long live FS!</description>
		<content:encoded><![CDATA[<p>You are wrong.  Asterisk sucks in many ways, it has flaws, I&#8217;ve tried to deploy it and found a million of bugs, which are already reported in mantis.</p>
<p>Not to mention that the SIP implementation in Asterisk sucks (chan_sip). whereas Sofia-SIP in FreeSWITCH is a lot better, and your comment about FreeSWITCH lacking in features is FUD.</p>
<p>Asterisk sucks, long live FS!</p>
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