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<DIV>Hi all,</DIV>
<DIV> </DIV>
<DIV>Just installed Asterisk 13 on CentOS 7 and have run into a problem.</DIV>
<DIV> </DIV>
<DIV>The Scenario is this:</DIV>
<DIV> </DIV>
<DIV>Asterisk is on the internet</DIV>
<DIV>the Phone, a D40, is behind NAT</DIV>
<DIV> </DIV>
<DIV>So someone calls the number and Asterisk routes the call to the D40</DIV>
<DIV>Everything works fine and the call is established, but then after 5 min. the caller stops getting audio from the D40 but there is still audio to the D40.</DIV>
<DIV> </DIV>
<DIV>using both RTP and SIP debug on the Asterisk console does not reveal anything.</DIV>
<DIV>Actually I can see from the RTP debug that RTP packages are send and received even after lose of the audio.</DIV>
<DIV> </DIV>
<DIV>So does anyone have any ideas where to look for the problem or perhaps a solution?</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV id=signature_old>
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<P style="FONT-FAMILY: Tahoma, Geneva, sans-serif"><SPAN lang=EN-US>Med venlig hilsen / Kind Regards,</SPAN></P>
<P style="FONT-FAMILY: Tahoma, Geneva, sans-serif"><SPAN lang=EN-US>Jonas Christoffersen</SPAN></P>
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<P><SPAN lang=DA style="FONT-FAMILY: Tahoma, Geneva, sans-serif">Tel. <SPAN style="WHITE-SPACE: nowrap">+45 3841 0960<BR></SPAN></SPAN><A style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma, Geneva, sans-serif; WHITE-SPACE: nowrap" href="http://www.showitmedia.eu/" target=_blank>www.showitmedia.eu</A><FONT face="Tahoma, Geneva, sans-serif"><BR></FONT><SPAN lang=DA style="FONT-SIZE: 10pt; FONT-FAMILY: arial, sans-serif"><SPAN style="WHITE-SPACE: nowrap"><A href="mailto:jonc@showitmedia.eu" target=_blank>jonc@showitmedia.eu</A></SPAN></SPAN><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: arial, sans-serif"> </SPAN></P>
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