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	<title>Comments on: 101 Things You Can Do With Asterisk</title>
	<atom:link href="http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/feed/" rel="self" type="application/rss+xml" />
	<link>http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/</link>
	<description>Open Source Software Engineering</description>
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		<title>By: Diego Viola</title>
		<link>http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/comment-page-1/#comment-210</link>
		<dc:creator>Diego Viola</dc:creator>
		<pubDate>Sat, 28 Jun 2008 19:02:40 +0000</pubDate>
		<guid isPermaLink="false">http://www.russellbryant.net/blog/?p=65#comment-210</guid>
		<description>if you rework the entire configuration interface please also allow regular expressions with PCRE for dialplans, etc.</description>
		<content:encoded><![CDATA[<p>if you rework the entire configuration interface please also allow regular expressions with PCRE for dialplans, etc.</p>
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	<item>
		<title>By: Diego Viola</title>
		<link>http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/comment-page-1/#comment-167</link>
		<dc:creator>Diego Viola</dc:creator>
		<pubDate>Wed, 18 Jun 2008 23:55:30 +0000</pubDate>
		<guid isPermaLink="false">http://www.russellbryant.net/blog/?p=65#comment-167</guid>
		<description>a socket interface will indeed rock... do you guys have plans for doing this in 1.6?</description>
		<content:encoded><![CDATA[<p>a socket interface will indeed rock&#8230; do you guys have plans for doing this in 1.6?</p>
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	<item>
		<title>By: Voip Enthusiast</title>
		<link>http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/comment-page-1/#comment-145</link>
		<dc:creator>Voip Enthusiast</dc:creator>
		<pubDate>Tue, 10 Jun 2008 23:39:33 +0000</pubDate>
		<guid isPermaLink="false">http://www.russellbryant.net/blog/?p=65#comment-145</guid>
		<description>Keep up the incredible job! =D</description>
		<content:encoded><![CDATA[<p>Keep up the incredible job! =D</p>
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	</item>
	<item>
		<title>By: Voip Enthusiast</title>
		<link>http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/comment-page-1/#comment-144</link>
		<dc:creator>Voip Enthusiast</dc:creator>
		<pubDate>Tue, 10 Jun 2008 23:39:06 +0000</pubDate>
		<guid isPermaLink="false">http://www.russellbryant.net/blog/?p=65#comment-144</guid>
		<description>Thank you for taking the time of answering my questions.

I really love asterisk and I think it&#039;s progressing rapidly.

I also look forward in your response to that guy =D</description>
		<content:encoded><![CDATA[<p>Thank you for taking the time of answering my questions.</p>
<p>I really love asterisk and I think it&#8217;s progressing rapidly.</p>
<p>I also look forward in your response to that guy =D</p>
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	<item>
		<title>By: russell</title>
		<link>http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/comment-page-1/#comment-143</link>
		<dc:creator>russell</dc:creator>
		<pubDate>Tue, 10 Jun 2008 22:36:24 +0000</pubDate>
		<guid isPermaLink="false">http://www.russellbryant.net/blog/?p=65#comment-143</guid>
		<description>&quot;I know you guys are doing the best to make the SIP stack better in Asterisk, adding new features, etc. and I’m really happy of that, but do you guys aim for RFC compliance?&quot;

Yes, of course we do.  However, that&#039;s not _all_ that we aim for.  In fact, our #1 priority is making it work with reality.  In reality, a lot of SIP implementations do weird things.  We make chan_sip act in weird ways sometimes to get the best possible interoperability that we can.  We have been extremely successful in that regard.

 &quot;I wonder if what this guy says about Asterisk is still true:

http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk&quot;

Most of that is not true anymore.  It&#039;s obvious to me that it was written by someone with knowledge about Asterisk 1.2 and earlier.  A _lot_ has changed since then.  Writing a detailed response to that article is on my to-do list.

&quot;See, I really like asterisk and if that’s the case I would like to see it improve, it is true that asterisk still depends on some modules to be able to start? is asterisk module independent now?&quot;

No, Asterisk does not depend on any modules to start.  The ones that were required have been moved into the core, which was only one of them (res_features).  In Asterisk 1.6, that code has moved into the core.</description>
		<content:encoded><![CDATA[<p>&#8220;I know you guys are doing the best to make the SIP stack better in Asterisk, adding new features, etc. and I’m really happy of that, but do you guys aim for RFC compliance?&#8221;</p>
<p>Yes, of course we do.  However, that&#8217;s not _all_ that we aim for.  In fact, our #1 priority is making it work with reality.  In reality, a lot of SIP implementations do weird things.  We make chan_sip act in weird ways sometimes to get the best possible interoperability that we can.  We have been extremely successful in that regard.</p>
<p> &#8220;I wonder if what this guy says about Asterisk is still true:</p>
<p><a href="http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk" rel="nofollow">http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk</a>&#8221;</p>
<p>Most of that is not true anymore.  It&#8217;s obvious to me that it was written by someone with knowledge about Asterisk 1.2 and earlier.  A _lot_ has changed since then.  Writing a detailed response to that article is on my to-do list.</p>
<p>&#8220;See, I really like asterisk and if that’s the case I would like to see it improve, it is true that asterisk still depends on some modules to be able to start? is asterisk module independent now?&#8221;</p>
<p>No, Asterisk does not depend on any modules to start.  The ones that were required have been moved into the core, which was only one of them (res_features).  In Asterisk 1.6, that code has moved into the core.</p>
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	<item>
		<title>By: Voip Enthusiast</title>
		<link>http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/comment-page-1/#comment-142</link>
		<dc:creator>Voip Enthusiast</dc:creator>
		<pubDate>Tue, 10 Jun 2008 21:04:50 +0000</pubDate>
		<guid isPermaLink="false">http://www.russellbryant.net/blog/?p=65#comment-142</guid>
		<description>&gt;5. A better development model? You may need to be more specific. git is a tool, not a development model.

You are right, Git is a tool... but it&#039;s a very innovative one, it&#039;s based on distribution (distributed development model).

I think you will really like it, and I don&#039;t meant to force you on using it, but you might want to check it out, it has some advantages.

Look this video ;-)

http://www.youtube.com/watch?v=4XpnKHJAok8

Git rocks!

&gt; 7. This is another blanket statement. Which features are you referring to, specifically? SIP TCP and TLS are already supported, and SRTP is coming soon.

I know you guys are doing the best to make the SIP stack better in Asterisk, adding new features, etc. and I&#039;m really happy of that, but do you guys aim for RFC compliance?</description>
		<content:encoded><![CDATA[<p>&gt;5. A better development model? You may need to be more specific. git is a tool, not a development model.</p>
<p>You are right, Git is a tool&#8230; but it&#8217;s a very innovative one, it&#8217;s based on distribution (distributed development model).</p>
<p>I think you will really like it, and I don&#8217;t meant to force you on using it, but you might want to check it out, it has some advantages.</p>
<p>Look this video <img src='http://www.russellbryant.net/blog/wp-includes/images/smilies/icon_wink.gif' alt=';-)' class='wp-smiley' /> </p>
<p><a href="http://www.youtube.com/watch?v=4XpnKHJAok8" rel="nofollow">http://www.youtube.com/watch?v=4XpnKHJAok8</a></p>
<p>Git rocks!</p>
<p>&gt; 7. This is another blanket statement. Which features are you referring to, specifically? SIP TCP and TLS are already supported, and SRTP is coming soon.</p>
<p>I know you guys are doing the best to make the SIP stack better in Asterisk, adding new features, etc. and I&#8217;m really happy of that, but do you guys aim for RFC compliance?</p>
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	<item>
		<title>By: Voip Enthusiast</title>
		<link>http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/comment-page-1/#comment-140</link>
		<dc:creator>Voip Enthusiast</dc:creator>
		<pubDate>Tue, 10 Jun 2008 19:41:58 +0000</pubDate>
		<guid isPermaLink="false">http://www.russellbryant.net/blog/?p=65#comment-140</guid>
		<description>If you rework the configuration interface, please don&#039;t get rid of the text configuration, I love asterisk configuration, I think plain text configuration is really fast, it could get better though...

That&#039;s one of the things I don&#039;t like about FS, that only XML is supported... and they lack some applications.

Asterisk is a lot better on this.</description>
		<content:encoded><![CDATA[<p>If you rework the configuration interface, please don&#8217;t get rid of the text configuration, I love asterisk configuration, I think plain text configuration is really fast, it could get better though&#8230;</p>
<p>That&#8217;s one of the things I don&#8217;t like about FS, that only XML is supported&#8230; and they lack some applications.</p>
<p>Asterisk is a lot better on this.</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Voip Enthusiast</title>
		<link>http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/comment-page-1/#comment-139</link>
		<dc:creator>Voip Enthusiast</dc:creator>
		<pubDate>Tue, 10 Jun 2008 19:39:57 +0000</pubDate>
		<guid isPermaLink="false">http://www.russellbryant.net/blog/?p=65#comment-139</guid>
		<description>I wonder if what this guy says about Asterisk is still true:

http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk

See, I really like asterisk and if that&#039;s the case I would like to see it improve, it is true that asterisk still depends on some modules to be able to start? is asterisk module independent now?

I don&#039;t have any reasons to switch to FS, I&#039;m used to asterisk and I like it, I see is getting better and better as time goes by... and I know 1.6 will rock as it gets more features and solid core.

Thanks for your hard work.</description>
		<content:encoded><![CDATA[<p>I wonder if what this guy says about Asterisk is still true:</p>
<p><a href="http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk" rel="nofollow">http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk</a></p>
<p>See, I really like asterisk and if that&#8217;s the case I would like to see it improve, it is true that asterisk still depends on some modules to be able to start? is asterisk module independent now?</p>
<p>I don&#8217;t have any reasons to switch to FS, I&#8217;m used to asterisk and I like it, I see is getting better and better as time goes by&#8230; and I know 1.6 will rock as it gets more features and solid core.</p>
<p>Thanks for your hard work.</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Voip Enthusiast</title>
		<link>http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/comment-page-1/#comment-136</link>
		<dc:creator>Voip Enthusiast</dc:creator>
		<pubDate>Tue, 10 Jun 2008 18:57:09 +0000</pubDate>
		<guid isPermaLink="false">http://www.russellbryant.net/blog/?p=65#comment-136</guid>
		<description>Keep up the good work.</description>
		<content:encoded><![CDATA[<p>Keep up the good work.</p>
]]></content:encoded>
	</item>
	<item>
		<title>By: Voip Enthusiast</title>
		<link>http://www.russellbryant.net/blog/2008/06/04/101-things-you-can-do-with-asterisk/comment-page-1/#comment-135</link>
		<dc:creator>Voip Enthusiast</dc:creator>
		<pubDate>Tue, 10 Jun 2008 18:56:48 +0000</pubDate>
		<guid isPermaLink="false">http://www.russellbryant.net/blog/?p=65#comment-135</guid>
		<description>Cool, thank you.

You are doing an awesome work with asterisk.</description>
		<content:encoded><![CDATA[<p>Cool, thank you.</p>
<p>You are doing an awesome work with asterisk.</p>
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