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	<title>Sekiur My Thoughts &#187; asterisk</title>
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	<link>http://blog.sekiur.com</link>
	<description>VoIP, Mobility, Security, Open Source, Science, Politics, and Technology.</description>
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		<title>PBX in a Flash with CBeyond</title>
		<link>http://blog.sekiur.com/2009/04/pbx-in-a-flash-with-cbeyond/</link>
		<comments>http://blog.sekiur.com/2009/04/pbx-in-a-flash-with-cbeyond/#comments</comments>
		<pubDate>Sun, 19 Apr 2009 16:57:11 +0000</pubDate>
		<dc:creator>Jose Vicente Ortega</dc:creator>
				<category><![CDATA[Technology]]></category>
		<category><![CDATA[aastra]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[cbeyond]]></category>
		<category><![CDATA[Google Voice]]></category>
		<category><![CDATA[open source]]></category>
		<category><![CDATA[Skype]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://blog.sekiur.com/?p=768</guid>
		<description><![CDATA[Last week I deployed a PBX in a Flash system using SIPConnect from CBeyond. It was so successful that I will start using PIAF in lieu of Trixbox from now on for all future deployments of this type and will replace my home PBX to take advantage of Skype and Google Voice integration. In this [...]]]></description>
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		<slash:comments>1</slash:comments>
		</item>
		<item>
		<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>
		</item>
		<item>
		<title>VoIP Phone Systems for Small Businesses</title>
		<link>http://blog.sekiur.com/2008/12/voip-phone-systems-for-small-businesses/</link>
		<comments>http://blog.sekiur.com/2008/12/voip-phone-systems-for-small-businesses/#comments</comments>
		<pubDate>Tue, 30 Dec 2008 05:10:33 +0000</pubDate>
		<dc:creator>Jose Vicente Ortega</dc:creator>
				<category><![CDATA[Business]]></category>
		<category><![CDATA[Technology]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[digium]]></category>
		<category><![CDATA[fonality]]></category>
		<category><![CDATA[grandstream]]></category>
		<category><![CDATA[jazinga]]></category>
		<category><![CDATA[microsoft]]></category>
		<category><![CDATA[response point]]></category>
		<category><![CDATA[rochbochs]]></category>
		<category><![CDATA[small business]]></category>
		<category><![CDATA[switchvox]]></category>
		<category><![CDATA[talkswitch]]></category>
		<category><![CDATA[trixbox]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://blog.sekiur.com/?p=666</guid>
		<description><![CDATA[The more I research on the potential and possibilities of VoIP phone systems, the more companies I see trying to get a piece of the market. Reminds me of a blog entry I read recently &#8220;Everything I Know About Business I Learned From Poker&#8221; and more specifically the quote: &#8220;If there are too many competitors [...]]]></description>
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		<slash:comments>0</slash:comments>
		</item>
		<item>
		<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>
		</item>
		<item>
		<title>Asterisk Success Story</title>
		<link>http://blog.sekiur.com/2008/11/asterisk-success-story/</link>
		<comments>http://blog.sekiur.com/2008/11/asterisk-success-story/#comments</comments>
		<pubDate>Fri, 14 Nov 2008 19:13:20 +0000</pubDate>
		<dc:creator>Jose Vicente Ortega</dc:creator>
				<category><![CDATA[Technology]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[microsoft]]></category>
		<category><![CDATA[sucess]]></category>
		<category><![CDATA[trixbox]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://blog.sekiur.com/?p=537</guid>
		<description><![CDATA[Just had to pass on what transpired today. I started a Tech Support call to Microsoft Partner Support at 9:05 this morning. The call was initially answered in Redmond by the Partner Group. It was then transferred (via IP) to India for First Level Support &#8211; this lasted for two hours, when it was kicked [...]]]></description>
		<wfw:commentRss>http://blog.sekiur.com/2008/11/asterisk-success-story/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Trixbox 2.6 and Sangoma Hardware</title>
		<link>http://blog.sekiur.com/2008/11/trixbox-26-and-sangoma-hardware/</link>
		<comments>http://blog.sekiur.com/2008/11/trixbox-26-and-sangoma-hardware/#comments</comments>
		<pubDate>Fri, 14 Nov 2008 18:37:40 +0000</pubDate>
		<dc:creator>Jose Vicente Ortega</dc:creator>
				<category><![CDATA[Technology]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[dell]]></category>
		<category><![CDATA[fonality]]></category>
		<category><![CDATA[nerd vittles]]></category>
		<category><![CDATA[open source]]></category>
		<category><![CDATA[piaf]]></category>
		<category><![CDATA[sangoma]]></category>
		<category><![CDATA[trixbox]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://blog.sekiur.com/?p=530</guid>
		<description><![CDATA[Trixbox (formerly Asterisk At Home &#8211; A@H) has definitely come a long since its beginnings in November 2004 and since I started playing around with Asterisk 2 months earlier. The convenience of being able to download an ISO and have a functional PBX in less than an hour was and is amazing. An excellent resource [...]]]></description>
		<wfw:commentRss>http://blog.sekiur.com/2008/11/trixbox-26-and-sangoma-hardware/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Hard SIP Phones &#8211; Cisco 7960 VoIP</title>
		<link>http://blog.sekiur.com/2005/07/hard-sip-phones-cisco-7960-voip/</link>
		<comments>http://blog.sekiur.com/2005/07/hard-sip-phones-cisco-7960-voip/#comments</comments>
		<pubDate>Sun, 03 Jul 2005 10:26:30 +0000</pubDate>
		<dc:creator>Jose Vicente Ortega</dc:creator>
				<category><![CDATA[Technology]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[cisco]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://blog.sekiur.com/?p=54</guid>
		<description><![CDATA[Received my Cisco 7960 SIP phone and althought it was tough to get the firmware (7.4) [P0S3-07-4-00], it was pretty straight forward to get the phone running. MORE&#8230; Basically the configuration had to be erased, the phone unlocked and pointed to the asterisk server where the tftp daemon is and the firmware lies. UPDATE: Run [...]]]></description>
		<wfw:commentRss>http://blog.sekiur.com/2005/07/hard-sip-phones-cisco-7960-voip/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Connecting Two Asterisk @Home Servers via IAX2</title>
		<link>http://blog.sekiur.com/2005/06/connecting-two-asterisk-home-servers-via-iax2/</link>
		<comments>http://blog.sekiur.com/2005/06/connecting-two-asterisk-home-servers-via-iax2/#comments</comments>
		<pubDate>Sat, 25 Jun 2005 22:06:26 +0000</pubDate>
		<dc:creator>Jose Vicente Ortega</dc:creator>
				<category><![CDATA[Technology]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[iax]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://blog.sekiur.com/?p=48</guid>
		<description><![CDATA[I have found very scattered and little documentation on connecting two asterisk servers via IAX2, so I decided to put these words together. IAX2 is a protocol that plays nice with NAT and very powerful. The most important thing to keep in mind is the client/server relationship when calling/forwarding from one server to another. So [...]]]></description>
		<wfw:commentRss>http://blog.sekiur.com/2005/06/connecting-two-asterisk-home-servers-via-iax2/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Broadvoice on Asterisk</title>
		<link>http://blog.sekiur.com/2005/06/broadvoice-on-asterisk/</link>
		<comments>http://blog.sekiur.com/2005/06/broadvoice-on-asterisk/#comments</comments>
		<pubDate>Mon, 06 Jun 2005 17:16:29 +0000</pubDate>
		<dc:creator>Jose Vicente Ortega</dc:creator>
				<category><![CDATA[Technology]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://blog.sekiur.com/?p=40</guid>
		<description><![CDATA[I signed up for a Broadvoice BYOD limited account which for about $6/month, I get 100 minutes of local calls and their great long distance rates&#8230;. at first it was a pain to set-up but this was probably because I was so tired&#8230; so I decided to give it one more try using instructions from [...]]]></description>
		<wfw:commentRss>http://blog.sekiur.com/2005/06/broadvoice-on-asterisk/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>FreeWorldDialUp Access Numbers</title>
		<link>http://blog.sekiur.com/2005/06/freeworlddialup-access-numbers/</link>
		<comments>http://blog.sekiur.com/2005/06/freeworlddialup-access-numbers/#comments</comments>
		<pubDate>Thu, 02 Jun 2005 13:09:24 +0000</pubDate>
		<dc:creator>Jose Vicente Ortega</dc:creator>
				<category><![CDATA[Technology]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[fwd]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://blog.sekiur.com/?p=38</guid>
		<description><![CDATA[The access numbers are as follows: Atlanta                678-918-5026 Atlantic City                609-840-8027 Baltimore                410-372-4173 Chicago                708-437-9040 Connecticut                860-256-4170 Dallas                469-449-2560 Fort Worth                817-886-2285 Daytona Beach                386-267-2650 Houston                832-631-1597 Los Angeles                323-908-4167 Miami                786-206-4270 New York                347-427-9019 N. New Jersey                973-494-5586 S. New Jersey                609-873-8119 Philadelphia                610-879-1419 San Francisco                415-354-1083 Seattle                206-219-5789 Washington, DC                202-742-5739 Spain +34 [...]]]></description>
		<wfw:commentRss>http://blog.sekiur.com/2005/06/freeworlddialup-access-numbers/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
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