This is a quite promising idea. Many thanks.<br>I'll post my results to the list...<br><br>Cheers,<br>--ag<br><br><div class="gmail_quote">On Fri, Jul 11, 2008 at 11:22 AM, Tilghman Lesher <<a href="mailto:tilghman@mail.jeffandtilghman.com">tilghman@mail.jeffandtilghman.com</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class="Ih2E3d">On Friday 11 July 2008 09:17:37 Artie Gold wrote:<br>
> In updating to 1.4.21 recently, we've encountered a problem, when running<br>
> over a satellite connection (where the latency is considerable; a "regular"<br>
> internet connection did not exhibit this problem), where incoming calls are<br>
> being dropped as a result of the sip handshake timing out (dropping down to<br>
> <a href="http://1.4.18.1" target="_blank">1.4.18.1</a> solved the problem for us). From reading the change logs and other<br>
> posts, it seems that some work has been done in this area recently to get<br>
> it "right"; it appears that, at least in the satellite case, things may<br>
> have gotten a little too "tight"...<br>
><br>
> If this rings a bell for anyone, any insight would be appreciated.<br>
<br>
</div>Try setting t1min to something higher than the default, 100 (ms). This value<br>
is settable globally, as well as per-peer.<br>
<br>
--<br>
<font color="#888888">Tilghman<br>
</font><div><div></div><div class="Wj3C7c"><br>
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