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	<title>Comments on: Simple VOIP test calls and Toll Fraud</title>
	<atom:link href="http://www.blindhog.net/simple-voip-test-calls-and-toll-fraud/feed/" rel="self" type="application/rss+xml" />
	<link>http://www.blindhog.net/simple-voip-test-calls-and-toll-fraud/</link>
	<description>Tips and Video Tutorials - Cisco .:. Linux .:. VOIP</description>
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		<title>By: Josh</title>
		<link>http://www.blindhog.net/simple-voip-test-calls-and-toll-fraud/comment-page-1/#comment-33861</link>
		<dc:creator>Josh</dc:creator>
		<pubDate>Sat, 07 Aug 2010 18:41:12 +0000</pubDate>
		<guid isPermaLink="false">http://www.blindhog.net/?p=908#comment-33861</guid>
		<description>Thanks for the link Scott!</description>
		<content:encoded><![CDATA[<p>Thanks for the link Scott!</p>
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		<title>By: Scott Newberry</title>
		<link>http://www.blindhog.net/simple-voip-test-calls-and-toll-fraud/comment-page-1/#comment-33860</link>
		<dc:creator>Scott Newberry</dc:creator>
		<pubDate>Sat, 07 Aug 2010 12:56:38 +0000</pubDate>
		<guid isPermaLink="false">http://www.blindhog.net/?p=908#comment-33860</guid>
		<description>Good stuff!  And take it from someone who has been burned...  Pay close attention to this!

The &quot;Security Best Practices&quot; section of the CME SRND should be required reading for all Cisco Voice engineers.  Not as much a concern when using MGCP, due to how the POTS trunks are controlled... Until you go into fallback.  So it IS still relevant!  Be careful out there!

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/security.html</description>
		<content:encoded><![CDATA[<p>Good stuff!  And take it from someone who has been burned&#8230;  Pay close attention to this!</p>
<p>The &#8220;Security Best Practices&#8221; section of the CME SRND should be required reading for all Cisco Voice engineers.  Not as much a concern when using MGCP, due to how the POTS trunks are controlled&#8230; Until you go into fallback.  So it IS still relevant!  Be careful out there!</p>
<p><a href="http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/security.html" rel="nofollow">http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/security.html</a></p>
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	<item>
		<title>By: Cisco: Mark voice packets at the network edge &#124; FirstDigest</title>
		<link>http://www.blindhog.net/simple-voip-test-calls-and-toll-fraud/comment-page-1/#comment-33539</link>
		<dc:creator>Cisco: Mark voice packets at the network edge &#124; FirstDigest</dc:creator>
		<pubDate>Tue, 04 May 2010 20:53:34 +0000</pubDate>
		<guid isPermaLink="false">http://www.blindhog.net/?p=908#comment-33539</guid>
		<description>[...] Simple VOIP test calls and Toll Fraud (blindhog.net) [...]</description>
		<content:encoded><![CDATA[<p>[...] Simple VOIP test calls and Toll Fraud (blindhog.net) [...]</p>
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	<item>
		<title>By: Josh</title>
		<link>http://www.blindhog.net/simple-voip-test-calls-and-toll-fraud/comment-page-1/#comment-33155</link>
		<dc:creator>Josh</dc:creator>
		<pubDate>Wed, 13 Jan 2010 07:07:18 +0000</pubDate>
		<guid isPermaLink="false">http://www.blindhog.net/?p=908#comment-33155</guid>
		<description>Ken,

If you restrict traffic between the CallManager and voice gateway, you will not be able to use the XLite SIP Phone.

You would need to modify the access-list or remove it temporarily.

Josh</description>
		<content:encoded><![CDATA[<p>Ken,</p>
<p>If you restrict traffic between the CallManager and voice gateway, you will not be able to use the XLite SIP Phone.</p>
<p>You would need to modify the access-list or remove it temporarily.</p>
<p>Josh</p>
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	</item>
	<item>
		<title>By: Josh</title>
		<link>http://www.blindhog.net/simple-voip-test-calls-and-toll-fraud/comment-page-1/#comment-33154</link>
		<dc:creator>Josh</dc:creator>
		<pubDate>Wed, 13 Jan 2010 07:05:13 +0000</pubDate>
		<guid isPermaLink="false">http://www.blindhog.net/?p=908#comment-33154</guid>
		<description>Sephiroth,

Not everyone will need to make changes to their gateway. It is just something to think about. 

Josh</description>
		<content:encoded><![CDATA[<p>Sephiroth,</p>
<p>Not everyone will need to make changes to their gateway. It is just something to think about. </p>
<p>Josh</p>
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		<title>By: Ken Johnson</title>
		<link>http://www.blindhog.net/simple-voip-test-calls-and-toll-fraud/comment-page-1/#comment-33150</link>
		<dc:creator>Ken Johnson</dc:creator>
		<pubDate>Tue, 12 Jan 2010 19:44:10 +0000</pubDate>
		<guid isPermaLink="false">http://www.blindhog.net/?p=908#comment-33150</guid>
		<description>Very interesting idea.  Thanks.

I&#039;ve always just used the &#039;csim start&#039; command.  This SIP method would allow an actual call.

I&#039;d have to think about the &#039;allow h323 to h323&#039;.  This would allow calls to forward off the VG.  Not sure if I want that.  But all I need is the &#039;allow sip to h323&#039; for the XLite SIP Phone to work.

Also, if you restrict traffic between the CallManager and voice gateway with the firewall ACL, will the XLite SIP phone work?  No SIP traffic from softphone to VG allowed.</description>
		<content:encoded><![CDATA[<p>Very interesting idea.  Thanks.</p>
<p>I&#8217;ve always just used the &#8216;csim start&#8217; command.  This SIP method would allow an actual call.</p>
<p>I&#8217;d have to think about the &#8216;allow h323 to h323&#8242;.  This would allow calls to forward off the VG.  Not sure if I want that.  But all I need is the &#8216;allow sip to h323&#8242; for the XLite SIP Phone to work.</p>
<p>Also, if you restrict traffic between the CallManager and voice gateway with the firewall ACL, will the XLite SIP phone work?  No SIP traffic from softphone to VG allowed.</p>
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		<title>By: Sephiroth</title>
		<link>http://www.blindhog.net/simple-voip-test-calls-and-toll-fraud/comment-page-1/#comment-33148</link>
		<dc:creator>Sephiroth</dc:creator>
		<pubDate>Tue, 12 Jan 2010 15:12:49 +0000</pubDate>
		<guid isPermaLink="false">http://www.blindhog.net/?p=908#comment-33148</guid>
		<description>Howdy Josh,

You are completely right man. It worked flawlessly. Since the VGW I managed has all its channels for voice I think I don´t need to put any restriction, only the restrictions for dialing.

Anyway, thanks a lot for the tip.

Thanks,
Sephiroth</description>
		<content:encoded><![CDATA[<p>Howdy Josh,</p>
<p>You are completely right man. It worked flawlessly. Since the VGW I managed has all its channels for voice I think I don´t need to put any restriction, only the restrictions for dialing.</p>
<p>Anyway, thanks a lot for the tip.</p>
<p>Thanks,<br />
Sephiroth</p>
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