<div dir="ltr">Hi Steve,<div><br></div><div>I tried doing the setting in default.xml. I got the same experience. </div><div><br></div><div>Thanks,</div><div>Jason</div><div><br></div></div><div class="gmail_extra"><br><br>
<div class="gmail_quote">On Mon, Jan 20, 2014 at 8:47 AM, Jason Sia <span dir="ltr"><<a href="mailto:jsia18@gmail.com" target="_blank">jsia18@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">I will try to fix the NAT, will the auto nat feature fix this?</div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Jan 20, 2014 at 7:45 AM, Steve Kieu <span dir="ltr"><<a href="mailto:msh.computing@gmail.com" target="_blank">msh.computing@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc 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><variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/><br>
</div><div><br></div><div>Or maybe better in many cases</div><div><br></div><div><variable name="sip-force-contact" value="NDLB-connectile-dysfunction-2.0"/><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"><<a href="mailto:jsia18@gmail.com" target="_blank">jsia18@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc 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'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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