<p>Does not work for me!</p>
<div class="gmail_quote">El 26/04/2012 20:14, "Carlos Alvarez" <<a href="mailto:carlos@televolve.com">carlos@televolve.com</a>> escribió:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Apr 26, 2012 at 10:47 AM, Danny Dias <span dir="ltr"><<a href="mailto:ing.diasdanny@gmail.com" target="_blank">ing.diasdanny@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p>I cant put public ip adress to the asterisk server.</p>
<p>The main problem i see is with the sip headers (contact, sdp ip and ports, etc)...with the result of one way audio or no registrstion at all. Does an STUN server should works?<br>
AGAIN: phones in one site behind nat and PBX in another site also behind nat</p></blockquote><div><br></div><div>No problem. I do that all the time. That's what I said in my message.</div><div><br></div></div>-- <br>
<div>Carlos Alvarez</div><div>TelEvolve</div><div><a href="tel:602-889-3003" value="+16028893003" target="_blank">602-889-3003</a></div><div><br></div><br>
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