<div>If your Astersik server behind NAT too, your need modify SIP.conf like this....</div>
<div> </div>
<div>externalIP= x.x.x.x</div>
<div>localnet= x.x.x.</div>
<div> </div>
<div>hope this can help you....<br><br> </div>
<div><span class="gmail_quote">On 12/8/05, <b class="gmail_sendername">Moises Silva</b> <<a href="mailto:moises.silva@gmail.com">moises.silva@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">what type of NAT do you have? sync? full cone? cone restricted, port restricted?<br>any messages in asterisk verbose console?
<br><br>best regards
<div><span class="e" id="q_10807a8970541f22_1"><br><br>
<div><span class="gmail_quote">On 12/7/05, <b class="gmail_sendername">chawki hammoud</b> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:cyhammoud@yahoo.com" target="_blank">cyhammoud@yahoo.com
</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">Hi list:<br>i have an asterisk box behind the NAT ,when i try to<br>send calls through Sip to the voip provider server the
<br>call is answered but in a one way calling,I hear the<br>voice of the other side just for 4 seconds and then <br>stop but the call do not hangup.<br><br>my sip.conf is:<br>[voip provider]<br>type=peer<br>host=<a onclick="return top.js.OpenExtLink(window,event,this)" href="http://213.112.50.8/" target="_blank">
213.112.50.8</a><br>username=XXXXXXX<br>secret=XXXXXX<br>fromuser=XXXXXXX<br>canreinvite=no <br>nat=yes<br>insercure=invite<br>disallow=all<br>allow=gsm<br><br><br><br><br>__________________________________________<br>Yahoo! DSL – Something to write home about.
<br>Just $16.99/mo. or less.<br><a onclick="return top.js.OpenExtLink(window,event,this)" href="http://dsl.yahoo.com/" target="_blank">dsl.yahoo.com</a><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by
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