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<div class="moz-cite-prefix">Am 03.11.2014 um 13:28 schrieb Tom
Braarup Cuykens:<br>
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<blockquote cite="mid:545774DD.2010502@plustel.dk" type="cite">First
I am new to PBX so i might be doing something fundamentally
wrong...
<br>
That being said I got a FreePBX 32bit stable 6.12.65.
<br>
<br>
I am having some issue with the NAT and sound, both phones are
ringing but there is sound, I had some talk on IRC:
<br>
<br>
<[TK]D-Fender> Note for elfranne's situation, :
nat=force_rport,comedia" should have returned the public IP the
call arrived on, but it is not. Can anyone comment on why it
wouldn't have pulled it?
<br>
<br>
A call sample 202 calling 203 (ignore 403):
<a class="moz-txt-link-freetext" href="http://pastebin.com/sPB6FJEu">http://pastebin.com/sPB6FJEu</a>
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<br>
<br>
<br>
<br>
</blockquote>
Hi Tom,<br>
<br>
you can add a STUN Server in your Linksys/SPA942-6.1.5(a) user
agents.<br>
<br>
read more about STUN at: <a class="moz-txt-link-freetext" href="http://www.voip-info.org/wiki/view/STUN">http://www.voip-info.org/wiki/view/STUN</a><br>
and there is a list of public STUN Server.<br>
<br>
Regards<br>
<br>
<div class="moz-signature">-- <br>
<b>Rainer Piper</b>
<br>
Integration engineer
<br>
Koeslinstr. 56
<br>
53123 BONN <br>
GERMANY
<br>
Phone: +49 228 97167161
<br>
P2P: <a class="moz-txt-link-freetext" href="sip:rainer@sip.soho-piper.de:5072">sip:rainer@sip.soho-piper.de:5072</a> (pjsip-test)
<br>
XMPP: <a class="moz-txt-link-abbreviated" href="mailto:rainer@xmpp.soho-piper.de">rainer@xmpp.soho-piper.de</a></div>
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