does thia apply on SIP only or also IAX?<br><br><div><span class="gmail_quote">On 6/5/06, <b class="gmail_sendername">Kevin P. Fleming</b> <<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>----- Osama Kamal <<a href="mailto:okamalo@gmail.com">okamalo@gmail.com</a>> wrote:
<br>> I am running asterisk behind nat, and 2 sip phones on 2 different adsl<br>> neted connections, asterisk is staying always in rtp media path, while<br>> canreinvite=yes is configured in both extensions. I need asterisk to
<br>> stay away from the rtp media path, what is wrong with that setup?<br><br>It
is nearly impossible to get a direct media path between two endpoints
that are both behind NATs, regardless of the SIP server/proxy you use.
Asterisk is no different in this regard.<br><br>--<br>Kevin P. Fleming<br>Senior Software Engineer<br>Digium, Inc.<br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">
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