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	<title>Sekiur My Thoughts &#187; sip</title>
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	<description>VoIP, Mobility, Security, Open Source, Science, Politics, and Technology.</description>
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		<title>Checkpoint Firewall-1 and The SIP Protocol</title>
		<link>http://blog.sekiur.com/2008/12/checkpoint-firewall-1-and-the-sip-protocol/</link>
		<comments>http://blog.sekiur.com/2008/12/checkpoint-firewall-1-and-the-sip-protocol/#comments</comments>
		<pubDate>Tue, 30 Dec 2008 06:58:38 +0000</pubDate>
		<dc:creator>Jose Vicente Ortega</dc:creator>
				<category><![CDATA[Security]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[checkpoint]]></category>
		<category><![CDATA[firewall-1]]></category>
		<category><![CDATA[fw-1]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://blog.sekiur.com/?p=669</guid>
		<description><![CDATA[You have an asterisk based VoIP phone system sitting on an internal network and you are trying to establish connectivity to a SIP-based trunk provider. You configure a static NAT entry 1-to-1 for the asterisk box and allow the SIP (udp 5060) through the firewall, but SIP registration fails constantly. While troubleshooting the issue you [...]]]></description>
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		<slash:comments>6</slash:comments>
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		<title>Monitoring VoIP Trunks</title>
		<link>http://blog.sekiur.com/2008/12/monitoring-voip-trunks/</link>
		<comments>http://blog.sekiur.com/2008/12/monitoring-voip-trunks/#comments</comments>
		<pubDate>Tue, 23 Dec 2008 03:28:19 +0000</pubDate>
		<dc:creator>Jose Vicente Ortega</dc:creator>
				<category><![CDATA[Technology]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[iax]]></category>
		<category><![CDATA[monitoring]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://blog.sekiur.com/?p=662</guid>
		<description><![CDATA[Using VoIP lines to save on long distance and/or international calls is smart but real savings come in when you are able to dump your landline and go all the way with VoIP. Over the years the technology has matured to the point where its possible to provide reliable phone service over the Internet. Vonage [...]]]></description>
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		<slash:comments>0</slash:comments>
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		<title>What is the problem with SIP &amp; NAT</title>
		<link>http://blog.sekiur.com/2005/05/what-is-the-problem-with-sip-nat/</link>
		<comments>http://blog.sekiur.com/2005/05/what-is-the-problem-with-sip-nat/#comments</comments>
		<pubDate>Wed, 18 May 2005 18:33:30 +0000</pubDate>
		<dc:creator>Jose Vicente Ortega</dc:creator>
				<category><![CDATA[Technology]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[firewall]]></category>
		<category><![CDATA[nat]]></category>
		<category><![CDATA[sip]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://blog.sekiur.com/?p=30</guid>
		<description><![CDATA[Getting SIP to work across a NAT or several of them from point to point can be very painful&#8230;. a sniffer and logs will help you greatly, if you can understand what is going on&#8230;&#8230; A SIP call involves both SIP signaling and RTP audio streams with the RTCP control streams as a hang-around. MORE&#8230; [...]]]></description>
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