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	<title>Excalibur Partners &#187; Asterisk</title>
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	<description>Technical References</description>
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		<title>Asterisk Called Party Name</title>
		<link>http://www.excaliburtech.net/archives/183</link>
		<comments>http://www.excaliburtech.net/archives/183#comments</comments>
		<pubDate>Sat, 03 Jul 2010 02:16:39 +0000</pubDate>
		<dc:creator>Admin</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Linux]]></category>
		<category><![CDATA[VOIP]]></category>
		<category><![CDATA[FreePBX]]></category>
		<category><![CDATA[Polycom]]></category>

		<guid isPermaLink="false">http://www.excaliburtech.net/?p=183</guid>
		<description><![CDATA[With a traditional PBX when dialing an extension the name is displayed on the phone. With Asterisk 1.4 and 1.6.x the name is not displayed even though phones, like the Polycom SoundPoint series, have support for it with the Remote-Party-ID SIP header. On the Asterisk issue tracker there was a proposed patch for Asterisk 1.2. [...]]]></description>
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		<title>Polycom Soft Keys and Exchange UM</title>
		<link>http://www.excaliburtech.net/archives/162</link>
		<comments>http://www.excaliburtech.net/archives/162#comments</comments>
		<pubDate>Sat, 06 Mar 2010 21:32:09 +0000</pubDate>
		<dc:creator>Admin</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Linux]]></category>
		<category><![CDATA[VOIP]]></category>
		<category><![CDATA[Exchange UM]]></category>
		<category><![CDATA[Polycom]]></category>

		<guid isPermaLink="false">http://www.excaliburtech.net/?p=162</guid>
		<description><![CDATA[In Asterisk 1.6, FreePBX, and Exchange UM I mentioned how to setup an extension to transfer a caller to voicemail. Let&#8217;s take this one step further and add a soft key on the Polycom. Instead of having to dial ##407ext, you will be able to press the Xfer VM button and it will prompt for [...]]]></description>
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		</item>
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		<title>Polycom BLF with Astersisk</title>
		<link>http://www.excaliburtech.net/archives/147</link>
		<comments>http://www.excaliburtech.net/archives/147#comments</comments>
		<pubDate>Sat, 06 Mar 2010 21:06:13 +0000</pubDate>
		<dc:creator>Admin</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Linux]]></category>
		<category><![CDATA[VOIP]]></category>
		<category><![CDATA[FreePBX]]></category>
		<category><![CDATA[Polycom]]></category>

		<guid isPermaLink="false">http://www.excaliburtech.net/?p=147</guid>
		<description><![CDATA[They are two methods of setting up BLF on Polycom phones. The first gives you idle or inuse status only. A ringing phone will show up with a solid inuse light. The good thing about this method is users can create BLF for extensions themselves. If this is all you need start by enabling the [...]]]></description>
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		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Asterisk 1.6, FreePBX and Exchange UM</title>
		<link>http://www.excaliburtech.net/archives/130</link>
		<comments>http://www.excaliburtech.net/archives/130#comments</comments>
		<pubDate>Mon, 01 Mar 2010 04:26:12 +0000</pubDate>
		<dc:creator>Admin</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Linux]]></category>
		<category><![CDATA[VOIP]]></category>
		<category><![CDATA[Exchange UM]]></category>
		<category><![CDATA[FreePBX]]></category>

		<guid isPermaLink="false">http://www.excaliburtech.net/?p=130</guid>
		<description><![CDATA[Setting up Asterisk 1.6.1 and FreePBX 2.5 or 2.6 to work with Exchange 2007 UM is easier than Asterisk 1.4 thanks to Asterisk 1.6 including support for SIP over TCP. However only Asterisk 1.6.1.4 and lower work without modification. The ability to redirect according to Moved Temporarily response from Exchange UM broken in version 1.6.1.5, [...]]]></description>
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		<item>
		<title>Asterisk and FreePBX Caller Initiated Blind Transfers</title>
		<link>http://www.excaliburtech.net/archives/126</link>
		<comments>http://www.excaliburtech.net/archives/126#comments</comments>
		<pubDate>Mon, 01 Mar 2010 02:54:51 +0000</pubDate>
		<dc:creator>Admin</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Linux]]></category>
		<category><![CDATA[VOIP]]></category>
		<category><![CDATA[FreePBX]]></category>

		<guid isPermaLink="false">http://www.excaliburtech.net/?p=126</guid>
		<description><![CDATA[If you place a call and perform a blind transfer Asterisk will continue dialplan execution with a result of ANSWER. This causes an issue with FreePBX as it thinks dialing on the trunk failed. The transfer will go through, but you receive the all circuits are busy message. The first time I reported this issue [...]]]></description>
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