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	<title>AudioCodes Forums</title>
	<link>http://audiocodes.websitetoolbox.com</link>
	<description>AudioCodes Forums</description>
	<ttl>60</ttl>
	<pubDate>Thur, 09 Feb 2012 21:37:05 GMT</pubDate>
	<item>
		<title>MP-118 cannot outcall</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5687000</link>
		<description>I cannot get the outcalls to work. When I dial out the call gets answered by the gateway but does not dial to the originating dialed number through one of the pstn lines.&lt;br&gt;&lt;br&gt;&lt;br&gt;&lt;br&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
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		<pubDate>Tue, 31 Jan 2012 13:52:26 GMT</pubDate>
		<author>John</author>
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	<item>
		<title>Mediant 1000 - Busy out PRI</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5659044</link>
		<description>Does anyone have a method of busying out the voice channels of a PRIs that does not terminate active calls?&amp;nbsp; I'm looking for a way to quiesce the PRIs so the Mediant can be taken out of service, rebooted, etc.&lt;br&gt;&lt;br&gt;&lt;br&gt;&lt;br&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
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		<pubDate>Thur, 12 Jan 2012 14:32:21 GMT</pubDate>
		<author>NIS</author>
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	<item>
		<title>AudioCode MP114 Caller ID</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5594006</link>
		<description>hi,&lt;br&gt;I got an audiocode that I configure with my asterix server. every thing is ok but when I receive a call for external it did not display the caller id of the external caller but the number of the sip count that I introduce in the Audiocode. how can I fix it?? please help me&lt;br&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
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		<pubDate>Thur, 24 Nov 2011 15:23:05 GMT</pubDate>
		<author>saigone</author>
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		<title>MP118 WebInterface stops at &quot;Loading in Progress&quot; after Firmware Upgrade</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5488627</link>
		<description>&lt;P&gt;Hello everybody,&lt;BR&gt;&lt;BR&gt;I've run into a problem while upgrading the firmware on our MP118s. On the first device, everything went smoothly when upgrading to&amp;nbsp;Firmware 6.20A.&lt;BR&gt;&lt;BR&gt;On the backup device I did upgrade to Firmware 6.20A via the update wizard. I uploaded the Firmware&amp;nbsp;cmp file and kept the rest&amp;nbsp;(configuration file etc.) in the original state - just like with the previous device.&lt;BR&gt;&lt;BR&gt;Once the firmware was burnt into the flash the device rebooted. When I tried to connect to the web interface, i got prompted for username and password and now the device keeps displaying the &quot;Loading In Progress...&quot; message indefinitely.&lt;BR&gt;&lt;BR&gt;I tried a factory reset, which changed the IP address, username and password&amp;nbsp;back to default - the problem with the web interface remains however.&lt;BR&gt;&lt;BR&gt;I'd appreciate any suggestions,&lt;/P&gt;Best regards,&lt;BR&gt;David&lt;BR&gt;&lt;br&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
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		<pubDate>Tue, 13 Sep 2011 07:01:50 GMT</pubDate>
		<author>David Faber</author>
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	<item>
		<title>Tandem Switching Mediant 2000</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5410294</link>
		<description>Hi Everyone,&lt;div&gt;&lt;br&gt;&lt;/div&gt;&lt;div&gt;Has anyone configured how to do tandem switching on Mediant?&lt;/div&gt;&lt;div&gt;&lt;br&gt;&lt;/div&gt;&lt;div&gt;&lt;br&gt;&lt;/div&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
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		<pubDate>Mon, 25 Jul 2011 08:02:14 GMT</pubDate>
		<author>Mike</author>
	</item>

	<item>
		<title>MP-114 wrong dial out</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5395548</link>
		<description>&lt;P&gt;&lt;SPAN style=&quot;FONT-FAMILY: 'Verdana','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US&quot; lang=EN-US&gt;Hi, I'm having a wired problem. For lab testing and demo purposing I set up a MP-114 FXO GW with Lync 2010. Everything went well until we tried to dial out to PTSN.&lt;/SPAN&gt;&lt;/P&gt;&lt;SPAN style=&quot;FONT-FAMILY: 'Verdana','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US&quot; lang=EN-US&gt;Calls cannot be completed except those beginning with 5 (five). Actually I realized that the GW is changing the first digit somehow, so I get a wrong number or a busy signal. I'm not making any digits manipulation. In the LOG I see the special digit &quot;e&quot; automatically added to the destination phone number:&lt;/SPAN&gt;&lt;BR&gt;&lt;SPAN style=&quot;FONT-FAMILY: 'Verdana','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US&quot; lang=EN-US&gt;NOTICE&amp;nbsp; : (&amp;nbsp;&amp;nbsp;&amp;nbsp;&amp;nbsp;&amp;nbsp; lgr_flow)(483&amp;nbsp;&amp;nbsp;&amp;nbsp;&amp;nbsp;&amp;nbsp;&amp;nbsp; )&amp;nbsp; |&amp;nbsp;&amp;nbsp;&amp;nbsp;&amp;nbsp;&amp;nbsp;&amp;nbsp; #0&lt;img src=&quot;/images/boards/smilies/biggrin.gif&quot; border=&quot;0&quot; align=&quot;absmiddle&quot;&gt;ialPhoneNumber=e42957024 &lt;BR&gt;NOTICE&amp;nbsp; : (&amp;nbsp;&amp;nbsp; lgr_psbrdif)(482&amp;nbsp;&amp;nbsp;&amp;nbsp;&amp;nbsp;&amp;nbsp;&amp;nbsp; )&amp;nbsp; #0 Sending DTMFs (e42957024, Dir=2, Len=9, InterTime=100, DigitLen=100)&lt;BR&gt;&lt;/SPAN&gt;&lt;BR&gt;&lt;SPAN style=&quot;FONT-FAMILY: 'Verdana','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US&quot; lang=EN-US&gt;Behavior changes if I disable the &lt;/SPAN&gt;&lt;SPAN style=&quot;FONT-FAMILY: 'Arial-BoldMT','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US; mso-bidi-font-family: Arial-BoldMT; mso-bidi-font-weight: bold&quot; lang=EN-US&gt;EnableMicrosofExt option. In that case, no call completes. &lt;/SPAN&gt;&lt;BR&gt;&lt;P&gt;&lt;SPAN style=&quot;FONT-FAMILY: 'Arial-BoldMT','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US; mso-bidi-font-family: Arial-BoldMT; mso-bidi-font-weight: bold&quot; lang=EN-US&gt;I tried several parameters changes, always getting same result. &lt;/SPAN&gt;&lt;/P&gt;&lt;P&gt;&lt;SPAN style=&quot;FONT-FAMILY: 'Arial-BoldMT','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US; mso-bidi-font-family: Arial-BoldMT; mso-bidi-font-weight: bold&quot; lang=EN-US&gt;Your help will be really appreciated. Thanks in advance. &lt;/SPAN&gt;&lt;/P&gt;&lt;P style=&quot;MARGIN: 0cm 0cm 0pt&quot; class=MsoNormal&gt;&lt;SPAN style=&quot;COLOR: #1f497d; mso-ansi-language: EN-US; mso-ascii-font-family: Calibri; mso-hansi-font-family: Calibri; mso-bidi-font-family: Calibri&quot; lang=EN-US&gt;&lt;FONT size=3 face=Calibri&gt;&amp;nbsp;&lt;/FONT&gt;&lt;/SPAN&gt;&lt;/P&gt;&lt;P&gt;&amp;nbsp;&lt;/P&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
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		<pubDate>Fri, 15 Jul 2011 19:57:14 GMT</pubDate>
		<author>Horacio Silva</author>
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	<item>
		<title>Monitoring ATA-MP114 Registered status through SNMP</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5354670</link>
		<description>Greetings,&lt;br&gt;&lt;br&gt;I am using ATA MP-114 and I configured them to use SIP and to register per endpoint.&lt;br&gt;&lt;br&gt;I am trying to monitor the endpoints' registration status.&lt;br&gt;It is very easy on the HTTP interface, but I can't find which OID to use to get this information through SNMP.&lt;br&gt;&lt;br&gt;Does anybody know which OID to use ?&lt;br&gt;&lt;br&gt;Thank you for your support.&lt;br&gt;&lt;br&gt;Regards,&lt;br&gt;DOUILLE Tristan&lt;br&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
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		<pubDate>Wed, 15 Jun 2011 15:28:46 GMT</pubDate>
		<author>Tristan Douille</author>
	</item>

	<item>
		<title>mediant 1000 no incoming call</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5268766</link>
		<description>Hi,&lt;div&gt;&lt;br&gt;&lt;/div&gt;&lt;div&gt;I just want to ask if somebody can tell me how to configure incoming calls?&lt;/div&gt;&lt;div&gt;I have a mediant 1000 with 1 Trunk card and 1x4 FXS card. Outgoing call has no problem, but I can't figure out how to obtain an incoming call.&lt;/div&gt;&lt;div&gt;&lt;br&gt;&lt;/div&gt;&lt;div&gt;I tried a lot of manipulations but non of it worked.&lt;/div&gt;&lt;div&gt;I always got &quot;Abnormal Disconnect cause:21#GWAPP_CALL_REJECTED Trunk:0 Conn:255&quot;.&lt;/div&gt;&lt;div&gt;&lt;br&gt;&lt;/div&gt;&lt;div&gt;Attached is the topology, Configuration, and some test logs.&lt;/div&gt;&lt;div&gt;The telephone number of E1 is 8587500 ISDN EURO.&lt;/div&gt;&lt;div&gt;&lt;br&gt;&lt;/div&gt;&lt;div&gt;Hope somebody can help me.&lt;/div&gt;&lt;div&gt;&lt;br&gt;&lt;/div&gt;&lt;div&gt;Thanks.&lt;/div&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5268766</guid>
		<pubDate>Sun, 22 May 2011 07:40:29 GMT</pubDate>
		<author>Mayks G.</author>
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	<item>
		<title>MP-202 B F2XS: SIP register isn't automatic</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5260000</link>
		<description>Hello,&lt;br&gt;I have a problem with the SIP registration on the ATA MP-202 B 2FXS&lt;br&gt;For example, after uploading my configuration file, the line of my ATA will have the status &quot;Not registred&quot; on &quot;Sip registration&quot;.&lt;br&gt;I must press OK on &quot;Signaling protocol&quot; tab on &quot;VoIP&quot; field to register my line.&lt;br&gt;I wish it would be done by itself.&lt;br&gt;How I can set up this?&lt;br&gt;For your information I'm editing directly the main conf file.&lt;br&gt;&lt;br&gt;Well it's a point but I have another problem :&lt;br&gt;In my conf file i chosed to unencrypt the password.&lt;br&gt;&lt;br&gt;So i use this command : (auth_password({toto})) on &quot;line parameters&quot; on &quot;VoIP&quot; field.&lt;br&gt;I found this command in the tutorial for the last software version (the one i'm using) but it doesn't work,&lt;br&gt;my CA doesn't recognize this password&lt;br&gt;&lt;br&gt;I checked the conf on MP202B interface and it shows me this : (auth_password(toto))&lt;br&gt;So I supose it understood that the password isn't encrypt but it not doing the hash by next.&lt;br&gt;&lt;br&gt;Just to clarify, the problem doesn't come from my CA because with the crypted password it's work.&lt;br&gt;&lt;br&gt;But I'm doing an Excel in which the customer can type the password he wants and that transcribe it in the line (auth_password({&quot;the_password_typed&quot;}))&lt;br&gt;&lt;br&gt;Even i have no way to crypt the password typed, can you tell me why the syntax {&quot;password&quot;} doesn't work for un-encrypt it ?&lt;br&gt;&lt;br&gt;I would be very grateful if someone could help me.&lt;br&gt;&lt;br&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5260000</guid>
		<pubDate>Tue, 17 May 2011 15:37:05 GMT</pubDate>
		<author>Jesson</author>
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	<item>
		<title>Audiocodes mp-118 will not save a deafult gateway</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5216302</link>
		<description>&lt;P&gt;By using the inbuilt voice control I can set the IP and the subnet mask, however, when i try to set the default gateway, after pressing 1 on the phone for &quot;save&quot; it ALWAYS resets to 0.0.0.0. Any ideas, or do I have a faulty device?&lt;/P&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5216302</guid>
		<pubDate>Wed, 04 May 2011 11:54:27 GMT</pubDate>
		<author>Daniel</author>
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	<item>
		<title>MP124 and TLS</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5200905</link>
		<description>Anyone have success using TLS on a MP-124?&amp;nbsp; If so, can you post the configuration settings you used?&amp;nbsp; Thanks.&lt;br&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5200905</guid>
		<pubDate>Sat, 23 Apr 2011 14:37:07 GMT</pubDate>
		<author>John</author>
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	<item>
		<title>MP-202B VLANs</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5152131</link>
		<description>Hello All!&lt;BR&gt;&lt;BR&gt;Does&amp;nbsp;MP-202B support VLANs?&lt;BR&gt;When I configure this device to use separate VLAN for data and voice only one of there is working (either a voice, or the data).&lt;BR&gt;&lt;BR&gt;When&lt;BR&gt;Data VLAN configured NAPT mode, metric = 1, default gateway checked and&lt;BR&gt;Voice&amp;nbsp;VLAN configured&amp;nbsp;ROUTE mode, metric = 0, default gateway checked only VoIP is working &lt;IMG border=0 align=absMiddle src=&quot;http://audiocodes.websitetoolbox.com/images/boards/smilies/confused.gif&quot;&gt;&lt;BR&gt;If for data VLAN set metric = 0 and for Voice VLAN set metric = 1 then data is working &lt;IMG border=0 align=absMiddle src=&quot;http://audiocodes.websitetoolbox.com/images/boards/smilies/mad.gif&quot;&gt;&lt;BR&gt;&lt;BR&gt;&lt;B&gt;What options to&amp;nbsp;set that it worked together?&lt;/B&gt;&lt;BR&gt;&lt;BR&gt;Thank you for information!&lt;BR&gt;&lt;BR&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
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		<pubDate>Wed, 23 Mar 2011 15:46:56 GMT</pubDate>
		<author>Yaroslav</author>
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	<item>
		<title>Set SIP Contact on a MP-112 FXS</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5149302</link>
		<description>Hi&lt;div&gt;&lt;br&gt;&lt;/div&gt;&lt;div&gt;I have a&amp;nbsp;MP-112 FXS where I need the to add the port number in the SIP reg Contact. Something like this:&lt;/div&gt;&lt;div&gt;Contact: &amp;lt;sip:xxxx@aaa.bbb.ccc.ddd:5060;transport=udp&amp;gt;&lt;/div&gt;&lt;div&gt;&lt;br&gt;&lt;/div&gt;&lt;div&gt;How du I do that?&lt;/div&gt;&lt;div&gt;&lt;br&gt;&lt;/div&gt;&lt;div&gt;Thanks&lt;/div&gt;&lt;div&gt;&lt;br&gt;&lt;/div&gt;&lt;div&gt;Lasse&lt;/div&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
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		<pubDate>Mon, 21 Mar 2011 21:29:26 GMT</pubDate>
		<author>Lasse Jorgensen</author>
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	<item>
		<title>Audiocodes Mediant 1000 : too short inter digit interval</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5141318</link>
		<description>&lt;DIV&gt;Hello,&lt;/DIV&gt;&lt;DIV&gt;&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;I have an Audiocodes Mediant 1000 VoIP gateway.&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;The gateway works fine in both sides (from SIP to PSTN) but when we send DTMF several digits didn't get from backward PSTNequipement , the gateway doesn't&amp;nbsp;send DTMF with the right interdigits value. &amp;nbsp;The backward equipment (with ISDN interconnexion) DTMF detector didn't get several digits because property &quot;inter digit interval&quot; which defines the time in millisecond between two tones seems too short, this needs to be at least 70ms. This value is actually about 38 ms. &lt;/DIV&gt;&lt;DIV&gt;&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;Configuration&lt;/DIV&gt;&lt;DIV&gt;========== &lt;/DIV&gt;&lt;DIV&gt;&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;Please find attached Board_valid.ini&lt;/DIV&gt;&lt;DIV&gt;&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;Codec : G_711&lt;/DIV&gt;&lt;DIV&gt;DTMF : RFC_2833 in SDP&lt;/DIV&gt;&lt;DIV&gt;&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;Your help is welcome&lt;/DIV&gt;&lt;DIV&gt;&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;Regards&lt;/DIV&gt;&lt;DIV&gt;&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;&amp;nbsp;&lt;/DIV&gt;&lt;DIV&gt;&lt;BR&gt;&lt;/DIV&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
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		<pubDate>Wed, 16 Mar 2011 14:33:40 GMT</pubDate>
		<author>Andria</author>
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	<item>
		<title>audiocodes mp-112 fax sending problem</title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5095771</link>
		<description>hy &lt;br&gt;&lt;br&gt;i've got an problem with my audiocodes mp-112 device while trying to send faxes. &lt;br&gt;the receiving of any fax is possible and that works fine!&lt;br&gt;if i try to send an fax over mp-112 device i'll get the following error in the error-log:&lt;br&gt;&lt;span id=&quot;dnn_ctr458_ContentPane&quot; class=&quot;DNNAlignleft&quot;&gt;&lt;span id=&quot;spBody&quot; class=&quot;Forum_Normal&quot;&gt;&lt;p&gt;4d:1h:25m:52s Invalid Tone Type (16). Channel ID:0     &lt;/p&gt; &lt;p&gt;4d:1h:25m:52s (   lgr_psbrdif)(4         ) !!   #0:failed to play tone&lt;/p&gt;&lt;p&gt;&lt;br&gt;&lt;/p&gt;&lt;p&gt;can anybody helps me? what does the &quot;failed to play tone&quot; message means?&lt;/p&gt;&lt;p&gt;thanks for your replys!&lt;br&gt;&lt;/p&gt;&lt;/span&gt;&lt;/span&gt;greetings...&lt;br&gt;lars&lt;br&gt;&lt;br&gt; &lt;p&gt;Forum: &lt;a href=&quot;http://audiocodes.websitetoolbox.com/?forum=128277&quot;&gt;Microsoft Forum&lt;/a&gt;
</description>
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		<pubDate>Wed, 16 Feb 2011 13:00:45 GMT</pubDate>
		<author>lars</author>
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