<div dir="ltr"><div class="gmail_extra"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div dir="ltr">I will try to fix the NAT, will the auto nat feature fix this?</div><div class="gmail_extra"><br></div></blockquote><div><br></div><div><br></div><div>auto-nat would automatically detect client behind nat and set the ext-rtp ip etc  correctly. But it can not tell the client to set the sip contact header right, it is client bug (notably linphone on desktop linux did it as I saw)</div>
<div>the best work around as I already posted is in FS (directory.xml for example add</div><div><br></div><div>&lt;variable name=&quot;sip-force-contact&quot; value=&quot;NDLB-connectile-dysfunction-2.0&quot;/&gt;<br></div>
<div><br></div><div><br></div><div><br></div><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div class="gmail_extra">
<br><div class="gmail_quote">On Mon, Jan 20, 2014 at 7:45 AM, Steve Kieu <span dir="ltr">&lt;<a href="mailto:msh.computing@gmail.com" target="_blank">msh.computing@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">Searching around the net you will find that usually the client does not handle Nat situation well - they put local IP in the sip contact. FS has a work around that using<div>

<br></div><div>&lt;variable name=&quot;sip-force-contact&quot; value=&quot;NDLB-connectile-dysfunction&quot;/&gt;<br>
</div><div><br></div><div>Or maybe better in many cases</div><div><br></div><div>&lt;variable name=&quot;sip-force-contact&quot; value=&quot;NDLB-connectile-dysfunction-2.0&quot;/&gt;<br></div><div><br></div><div>that should work for you.</div>


<div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div>On Mon, Jan 20, 2014 at 2:27 AM, Jason Sia <span dir="ltr">&lt;<a href="mailto:jsia18@gmail.com" target="_blank">jsia18@gmail.com</a>&gt;</span> wrote:<br>


</div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><div><div dir="ltr">Hi,<div><br></div><div>
My freeswitch is sending out a BYE signal due to ACK timeout even though there is already audio .  The audio gets cut off after around 30 seconds, I think it detects RTP timeout.  It&#39;s wierd because I already picked up the call and I can already do conversation but after 30 seconds it just cuts it off.  Is there any solution for this?  Im using freeswitch 1.5 btw.</div>



<div><br></div><div>Thanks,</div><div>Jason</div></div>
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<br></div></blockquote></div><span><font color="#888888"><br><br clear="all"><div><br></div>-- <br>Steve Kieu
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Steve Kieu
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