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	<title>AndyB &#187; asterisk</title>
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	<description>shakin' up noth-ink</description>
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		<title>Provisioning Linksys SPA9*2 Phones</title>
		<link>http://andrewbevitt.com/2009/01/03/provisioning-linksys-spa92-phones/</link>
		<comments>http://andrewbevitt.com/2009/01/03/provisioning-linksys-spa92-phones/#comments</comments>
		<pubDate>Sat, 03 Jan 2009 12:28:12 +0000</pubDate>
		<dc:creator>AndyB</dc:creator>
				<category><![CDATA[Work]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[bulk]]></category>
		<category><![CDATA[provision]]></category>
		<category><![CDATA[spa$PSN.cfg]]></category>
		<category><![CDATA[spa942]]></category>
		<category><![CDATA[spa962]]></category>
		<category><![CDATA[tftp]]></category>
		<category><![CDATA[voip]]></category>

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		<guid isPermaLink="false">http://andrewbevitt.com/?p=152</guid>
		<description><![CDATA[<br/>I&#8217;ve just posted a new tutorial on how to do Bulk Provisioning of Linksys SPA9*2 SIP desk phones with tftp using xml configuration files. This works for both the 5.2.8 and 6.1.3a firmware for both the SPA942 and SPA962 (with or without the side car).]]></description>
			<content:encoded><![CDATA[<br/><p>I&#8217;ve just posted a new tutorial on how to do <a href="http://andrewbevitt.com/tutorials/bulk-linksys-spa92-provisioning/">Bulk Provisioning of Linksys SPA9*2</a> SIP desk phones with tftp using xml configuration files. This works for both the 5.2.8 and 6.1.3a firmware for both the SPA942 and SPA962 (with or without the side car).</p>
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		<title>Asterisk 1.6 Segmentation Faults at VoiceMail</title>
		<link>http://andrewbevitt.com/2008/12/29/asterisk-16-segmentation-faults-at-voicemail/</link>
		<comments>http://andrewbevitt.com/2008/12/29/asterisk-16-segmentation-faults-at-voicemail/#comments</comments>
		<pubDate>Mon, 29 Dec 2008 11:20:18 +0000</pubDate>
		<dc:creator>AndyB</dc:creator>
				<category><![CDATA[Work]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[segfault]]></category>
		<category><![CDATA[voicemail]]></category>

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		<guid isPermaLink="false">http://andrewbevitt.com/?p=132</guid>
		<description><![CDATA[<br/>Just a brief note on segfaults when compiling Asterisk 1.6.0 (through to 1.6.0.3-rc1 and perhaps onwards). If you find that the Asterisk process segfaults when a caller starts to record a VoiceMail message then go back to the build step: make menuselect Compiler Flags -> DONT OPTIMIZE And then recompile / reinstall.]]></description>
			<content:encoded><![CDATA[<br/><p>Just a brief note on segfaults when compiling Asterisk 1.6.0 (through to 1.6.0.3-rc1 and perhaps onwards). If you find that the Asterisk process segfaults when a caller starts to record a VoiceMail message then go back to the build step:</p>
<p><code>make menuselect<br />
Compiler Flags -> DONT OPTIMIZE</code></p>
<p>And then recompile / reinstall.</p>
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