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	<title>Patrick's Blog(2) &#187; VoIP</title>
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	<link>http://blog.laimbock.com</link>
	<description>Observations in a fast changing World</description>
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		<title>VoIP client and IPtables port forwarding</title>
		<link>http://blog.laimbock.com/2010/04/04/voip-client-and-iptables-port-forwarding/</link>
		<comments>http://blog.laimbock.com/2010/04/04/voip-client-and-iptables-port-forwarding/#comments</comments>
		<pubDate>Sun, 04 Apr 2010 18:21:32 +0000</pubDate>
		<dc:creator>Patrick</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://blog.laimbock.com/?p=233</guid>
		<description><![CDATA[Let&#8217;s say you have an Asterisk server that listens on the standard UDP port 5060 and want to use the VoIP client on for example your shiny new Nokia N900. You configure the VoIP client and everything works as advertised via your local WiFi link.
Next you also want to use the 3G/3.5G Internet link on [...]]]></description>
		<wfw:commentRss>http://blog.laimbock.com/2010/04/04/voip-client-and-iptables-port-forwarding/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
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		<item>
		<title>FreeSWITCH 1.0.4 released</title>
		<link>http://blog.laimbock.com/2009/08/08/freeswitch-1-0-4-released/</link>
		<comments>http://blog.laimbock.com/2009/08/08/freeswitch-1-0-4-released/#comments</comments>
		<pubDate>Sat, 08 Aug 2009 13:44:40 +0000</pubDate>
		<dc:creator>Patrick</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://blog.laimbock.com/?p=177</guid>
		<description><![CDATA[As announced on FreeSWITCH&#8217;s frontpage, version 1.0.4 has been released and now supports Skype, ZRTP, H.323 and MRCP: The FreeSWITCH team is pleased to announce the immediately availability of FreeSWITCH version 1.0.4. (Source tarball available here.) This new version contains many improvements in stability and security as well as some notable additions. More information here [...]]]></description>
		<wfw:commentRss>http://blog.laimbock.com/2009/08/08/freeswitch-1-0-4-released/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
		</item>
		<item>
		<title>31 great free applications for Asterisk</title>
		<link>http://blog.laimbock.com/2009/08/08/31-great-free-applications-for-asterisk/</link>
		<comments>http://blog.laimbock.com/2009/08/08/31-great-free-applications-for-asterisk/#comments</comments>
		<pubDate>Sat, 08 Aug 2009 13:40:14 +0000</pubDate>
		<dc:creator>Patrick</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://blog.laimbock.com/?p=175</guid>
		<description><![CDATA[In case you are still using Asterisk and haven&#8217;t switched to FreeSWITCH yet, Venturevoip has a nice overview of 31 great free applications for Asterisk.
]]></description>
		<wfw:commentRss>http://blog.laimbock.com/2009/08/08/31-great-free-applications-for-asterisk/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Cisco 7961G SIP is buggy</title>
		<link>http://blog.laimbock.com/2008/07/08/cisco-7961g-sip-support-is-buggy/</link>
		<comments>http://blog.laimbock.com/2008/07/08/cisco-7961g-sip-support-is-buggy/#comments</comments>
		<pubDate>Tue, 08 Jul 2008 11:27:57 +0000</pubDate>
		<dc:creator>Patrick</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://blog.laimbock.com/?p=94</guid>
		<description><![CDATA[This week I updated the FreeSWITCH test box and decided to hook up a Cisco 7961G phone. To my surprise the 7961G would not register to the FreeSWITCH box. Further investigation showed that the 7961G with firmware 8.3.5 sends the following register to the FreeSWITCH box:
REGISTER sip:10.9.123.36 SIP/2.0
Via: SIP/2.0/UDP 10.4.11.25:5060;branch=z9hG4bK8b5670ec
From: &#60;sip:1000@10.9.123.36&#62;;tag=0016467650c80002ed42b648-ff2b3abc
To: &#60;sip:1000@10.9.123.36&#62;
Call-ID: 00164676-50c80002-86027558-17f7cc0c@10.4.11.25
Max-Forwards: 70
Date: Tue, [...]]]></description>
		<wfw:commentRss>http://blog.laimbock.com/2008/07/08/cisco-7961g-sip-support-is-buggy/feed/</wfw:commentRss>
		<slash:comments>5</slash:comments>
		</item>
		<item>
		<title>Encrypted VoIP calls are vulnerable</title>
		<link>http://blog.laimbock.com/2008/06/13/encrypted-voip-calls-are-vulnerable/</link>
		<comments>http://blog.laimbock.com/2008/06/13/encrypted-voip-calls-are-vulnerable/#comments</comments>
		<pubDate>Fri, 13 Jun 2008 16:47:55 +0000</pubDate>
		<dc:creator>Patrick</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://blog.laimbock.com/?p=89</guid>
		<description><![CDATA[New Scientist has an interesting article about the vulnerability of encrypted VoIP packets:  &#8220;Security researchers at Johns Hopkins report that a variable bit-rate compression scheme being rolled out on VoIP systems leaves encrypted calls vulnerable to bugging. Simpler syllables are squeezed into smaller data packets, with more complex ones taking up more space; the [...]]]></description>
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		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>FreeSWITCH 1.0.0 &#8220;Phoenix&#8221; Released!</title>
		<link>http://blog.laimbock.com/2008/05/27/freeswitch-100-phoenix-released/</link>
		<comments>http://blog.laimbock.com/2008/05/27/freeswitch-100-phoenix-released/#comments</comments>
		<pubDate>Tue, 27 May 2008 18:46:31 +0000</pubDate>
		<dc:creator>Patrick</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://blog.laimbock.com/2008/05/27/freeswitch-100-phoenix-released/</guid>
		<description><![CDATA[FreeSWITCH 1.0.0 &#8220;Phoenix&#8221; has been released. Congratulations to Anthony, Mike, Brian and everybody who has contributed to make this happen! For quite some time I have witnessed FS in the making on irc and the mailinglist. The amount of effort that has gone into creating FS is quite astonishing. It&#8217;s all too easy to take [...]]]></description>
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		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Asterisk 1.2 and Xs4all VoIP configuration</title>
		<link>http://blog.laimbock.com/2007/08/31/asterisk-12-and-xs4all-voip-configuration/</link>
		<comments>http://blog.laimbock.com/2007/08/31/asterisk-12-and-xs4all-voip-configuration/#comments</comments>
		<pubDate>Fri, 31 Aug 2007 13:11:44 +0000</pubDate>
		<dc:creator>Patrick</dc:creator>
				<category><![CDATA[Linux and Open Source]]></category>
		<category><![CDATA[Tips and tricks]]></category>
		<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://blog.laimbock.com/2007/08/31/asterisk-12-and-xs4all-voip-configuration/</guid>
		<description><![CDATA[A while back the excellent Dutch ISP Xs4all introduced a VoIP service for its subscribers. Here is the configuration to make Asterisk 1.2 work with Xs4all. But before we go into the config one remark. There is a problem with Asterisk and how the Cirpack switch that Xs4all uses sends DTMF tones. You have to [...]]]></description>
		<wfw:commentRss>http://blog.laimbock.com/2007/08/31/asterisk-12-and-xs4all-voip-configuration/feed/</wfw:commentRss>
		<slash:comments>2</slash:comments>
		</item>
		<item>
		<title>Inband DTMF detection on Asterisk 1.2 broken</title>
		<link>http://blog.laimbock.com/2007/08/31/inband-dtmf-detection-on-asterisk-12-broken/</link>
		<comments>http://blog.laimbock.com/2007/08/31/inband-dtmf-detection-on-asterisk-12-broken/#comments</comments>
		<pubDate>Fri, 31 Aug 2007 10:20:22 +0000</pubDate>
		<dc:creator>Patrick</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://blog.laimbock.com/2007/08/31/inband-dtmf-detection-on-asterisk-12-broken/</guid>
		<description><![CDATA[Tony Mountifield recently found out that the inband detection of DTMF in Asterisk 1.2 was broken when the tones were not perfect. Check the bug report here. Because Asterisk 1.2 is in security maintenance mode only, Tony&#8217;s fix will not be committed to the Asterisk 1.2 code tree. Either get the patch from the bugreport [...]]]></description>
		<wfw:commentRss>http://blog.laimbock.com/2007/08/31/inband-dtmf-detection-on-asterisk-12-broken/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>FreePBX release 2.3 available</title>
		<link>http://blog.laimbock.com/2007/08/29/freepbx-release-23-available/</link>
		<comments>http://blog.laimbock.com/2007/08/29/freepbx-release-23-available/#comments</comments>
		<pubDate>Wed, 29 Aug 2007 11:14:49 +0000</pubDate>
		<dc:creator>Patrick</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://blog.laimbock.com/2007/08/29/freepbx-release-23-available/</guid>
		<description><![CDATA[The FreePBX project has released version 2.3. This release fixes more than 250 bugs in 2.3 and earlier releases. It is also the first release to support Asterisk 1.4. Read more about FreePBX here and download FreePBX here
]]></description>
		<wfw:commentRss>http://blog.laimbock.com/2007/08/29/freepbx-release-23-available/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Mobile operators heading for price slash?</title>
		<link>http://blog.laimbock.com/2006/02/20/mobile-operators-heading-for-price-slash/</link>
		<comments>http://blog.laimbock.com/2006/02/20/mobile-operators-heading-for-price-slash/#comments</comments>
		<pubDate>Mon, 20 Feb 2006 15:40:32 +0000</pubDate>
		<dc:creator>Patrick</dc:creator>
				<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://blog.laimbock.com/?p=49</guid>
		<description><![CDATA[Microsoft&#8217;s Steve Ballmer of monkey dance fame showed an interesting little gadget at the 3GSM show in Barcelona. A Skype like VoIP application on a mobile phone running some Microsoft OS that uses WiFi to make free VoIP calls using the Internet. Read more at TheBusinessOnline.com.
]]></description>
		<wfw:commentRss>http://blog.laimbock.com/2006/02/20/mobile-operators-heading-for-price-slash/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
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