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<DIV><FONT face=Arial size=2>Hi Geoff,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>You might want to try tcdump, specifying the source
and destination IP (to minimize the info)</FONT></DIV>
<DIV><FONT face=Arial size=2>and see where are the RTP packets going ;
</FONT><FONT face=Arial size=2>you will see if they change port or
something like that</FONT></DIV>
<DIV><FONT face=Arial size=2>after a while.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Cheers,</FONT></DIV>
<DIV><FONT face=Arial size=2>Frederic</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=geoffm33@gmail.com href="mailto:geoffm33@gmail.com">Geoff Manning</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">Asterisk Users Mailing List -
Non-Commercial Discussion</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, April 25, 2006 17:37</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] One Way
Audio....in the middle of a call</DIV>
<DIV><BR></DIV>We had a user report that they were on a SIP <---> PSTN
call for about 4.5 minutes before the call went to on-way audio. The user
called the person back and they reported being able to hear my user, but my
user couldn't hear them. The audio condition persisted for about 15 seconds
before the user hung up. <BR><BR>Where do I start to troubleshoot one way
audio that occurs during a call?<BR>
<P>
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