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	<title>Comments for Asterisk Tutorials</title>
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	<link>http://www.asterisktutorials.com</link>
	<description>Free Video Tutorials for trixbox FreePBX and Asterisk Systems</description>
	<lastBuildDate>Wed, 03 Mar 2010 17:09:00 +0000</lastBuildDate>
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		<title>Comment on Xorcom Astribank Interface Overview by Phil</title>
		<link>http://www.asterisktutorials.com/xorcom-astribank-interface-overview/comment-page-1/#comment-1980</link>
		<dc:creator>Phil</dc:creator>
		<pubDate>Wed, 03 Mar 2010 17:09:00 +0000</pubDate>
		<guid isPermaLink="false">http://www.asterisktutorials.com/?p=88#comment-1980</guid>
		<description>nice presentation, these look very nice etc, and i can&#039;t wait for the tutorial and seeing how easy these are to set up and configure in your next video.</description>
		<content:encoded><![CDATA[<p>nice presentation, these look very nice etc, and i can&#8217;t wait for the tutorial and seeing how easy these are to set up and configure in your next video.</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on trixbox CE &#8211; Creating an inbound dialplan / IVR menu by greg</title>
		<link>http://www.asterisktutorials.com/trixbox-ce-creating-an-inbound-dialplan-ivr-menu/comment-page-1/#comment-1773</link>
		<dc:creator>greg</dc:creator>
		<pubDate>Mon, 15 Feb 2010 22:11:56 +0000</pubDate>
		<guid isPermaLink="false">http://www.asterisktutorials.com/?p=48#comment-1773</guid>
		<description>really great, thanks.</description>
		<content:encoded><![CDATA[<p>really great, thanks.</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on Configuring snom 3xx series phones with trixbox CE by admin</title>
		<link>http://www.asterisktutorials.com/configuring-snom-3xx-series-phones-with-trixbox-ce/comment-page-1/#comment-1339</link>
		<dc:creator>admin</dc:creator>
		<pubDate>Wed, 20 Jan 2010 18:35:59 +0000</pubDate>
		<guid isPermaLink="false">http://www.asterisktutorials.com/?p=69#comment-1339</guid>
		<description>Yes, virtually all phones support option 66 but they all tend to do it a little differently. Consult the administrators guide on provisioning for exactly what the DHCP server has to push out for your phone.</description>
		<content:encoded><![CDATA[<p>Yes, virtually all phones support option 66 but they all tend to do it a little differently. Consult the administrators guide on provisioning for exactly what the DHCP server has to push out for your phone.</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on Configuring snom 3xx series phones with trixbox CE by Andrew</title>
		<link>http://www.asterisktutorials.com/configuring-snom-3xx-series-phones-with-trixbox-ce/comment-page-1/#comment-1337</link>
		<dc:creator>Andrew</dc:creator>
		<pubDate>Wed, 20 Jan 2010 17:10:01 +0000</pubDate>
		<guid isPermaLink="false">http://www.asterisktutorials.com/?p=69#comment-1337</guid>
		<description>I didnt know about DHCP option 66, does grandstream phone support this?</description>
		<content:encoded><![CDATA[<p>I didnt know about DHCP option 66, does grandstream phone support this?</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on Using ENUM with FreePBX by admin</title>
		<link>http://www.asterisktutorials.com/using-enum-with-freepbx/comment-page-1/#comment-1219</link>
		<dc:creator>admin</dc:creator>
		<pubDate>Fri, 08 Jan 2010 18:01:47 +0000</pubDate>
		<guid isPermaLink="false">http://www.asterisktutorials.com/?p=8#comment-1219</guid>
		<description>Correct, if the call can be made directly, then it goes out over your internet connection and is handled PBX to PBX with no carrier in the middle.</description>
		<content:encoded><![CDATA[<p>Correct, if the call can be made directly, then it goes out over your internet connection and is handled PBX to PBX with no carrier in the middle.</p>
]]></content:encoded>
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	<item>
		<title>Comment on Using ENUM with FreePBX by N</title>
		<link>http://www.asterisktutorials.com/using-enum-with-freepbx/comment-page-1/#comment-1218</link>
		<dc:creator>N</dc:creator>
		<pubDate>Fri, 08 Jan 2010 16:50:14 +0000</pubDate>
		<guid isPermaLink="false">http://www.asterisktutorials.com/?p=8#comment-1218</guid>
		<description>This is great stuff; honestly!!
Although, does this mean you do not need to pay &quot;anything&quot; for calls between these two numbers?</description>
		<content:encoded><![CDATA[<p>This is great stuff; honestly!!<br />
Although, does this mean you do not need to pay &#8220;anything&#8221; for calls between these two numbers?</p>
]]></content:encoded>
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	<item>
		<title>Comment on Using ENUM with FreePBX by marek</title>
		<link>http://www.asterisktutorials.com/using-enum-with-freepbx/comment-page-1/#comment-1207</link>
		<dc:creator>marek</dc:creator>
		<pubDate>Thu, 07 Jan 2010 11:40:35 +0000</pubDate>
		<guid isPermaLink="false">http://www.asterisktutorials.com/?p=8#comment-1207</guid>
		<description>very nice tutorial, thanks!</description>
		<content:encoded><![CDATA[<p>very nice tutorial, thanks!</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on trixbox CE &#8211; Creating an inbound dialplan / IVR menu by daniel</title>
		<link>http://www.asterisktutorials.com/trixbox-ce-creating-an-inbound-dialplan-ivr-menu/comment-page-1/#comment-976</link>
		<dc:creator>daniel</dc:creator>
		<pubDate>Thu, 17 Dec 2009 09:14:06 +0000</pubDate>
		<guid isPermaLink="false">http://www.asterisktutorials.com/?p=48#comment-976</guid>
		<description>With regard to the issue of the audio not playing:

if you uploaded a wav file, this needs to be with the following attributes:
Bit Rate: 128
Audio Sample Size: 16 bit
Sample Rate: 8kHz

alternativley, record from you phone.

hope this helps.</description>
		<content:encoded><![CDATA[<p>With regard to the issue of the audio not playing:</p>
<p>if you uploaded a wav file, this needs to be with the following attributes:<br />
Bit Rate: 128<br />
Audio Sample Size: 16 bit<br />
Sample Rate: 8kHz</p>
<p>alternativley, record from you phone.</p>
<p>hope this helps.</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on Linksys SPA941/942 SIP Phones by Roel</title>
		<link>http://www.asterisktutorials.com/linksys-spa941942-sip-phones/comment-page-1/#comment-917</link>
		<dc:creator>Roel</dc:creator>
		<pubDate>Thu, 10 Dec 2009 01:48:47 +0000</pubDate>
		<guid isPermaLink="false">http://www.asterisktutorials.com/?p=16#comment-917</guid>
		<description>It really  help thank you</description>
		<content:encoded><![CDATA[<p>It really  help thank you</p>
]]></content:encoded>
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	<item>
		<title>Comment on Cisco 7970 IP Phone by zzbox</title>
		<link>http://www.asterisktutorials.com/cisco-7970-ip-phone/comment-page-1/#comment-912</link>
		<dc:creator>zzbox</dc:creator>
		<pubDate>Wed, 09 Dec 2009 11:03:59 +0000</pubDate>
		<guid isPermaLink="false">http://www.asterisktutorials.com/?p=15#comment-912</guid>
		<description>I try this on my model 7971-GE. but it picks up sipload term71.default.loads. What do I have to have in xmldefault to be able to pick up the correct sipload ?</description>
		<content:encoded><![CDATA[<p>I try this on my model 7971-GE. but it picks up sipload term71.default.loads. What do I have to have in xmldefault to be able to pick up the correct sipload ?</p>
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