Forward UDP Ports 10000-20000 to your asterisk box.<br>
<br><br><div><span class="gmail_quote">On 12/8/05, <b class="gmail_sendername">Jeffery Chen</b> <<a href="mailto:jeffery9@gmail.com">jeffery9@gmail.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;">
<div>can u paste your sip.conf general section,,?</div>
<div> </div>
<div>there have another possible cause... the both side use different codecm and asterisk can not translaste it ...</div>
<div> </div>
<div><br>-- <br><span class="sg">Jeffery <br><br> </span></div><div><span class="e" id="q_1080974a548ee272_3">
<div><span class="gmail_quote">On 12/8/05, <b class="gmail_sendername">chawki hammoud</b> <<a href="mailto:cyhammoud@yahoo.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">cyhammoud@yahoo.com
</a>> wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">Hi:<br>i added these two lines to my general context ,but<br>nothing happened the same result the sound came in one
<br>way for 3 seconds and stopped but it didnt hangup.<br><br>--- Jeffery Chen <<a href="mailto:jeffery9@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">jeffery9@gmail.com</a>> wrote:
<br><br>> If your Astersik server behind NAT too, your need<br>> modify
SIP.conf like<br>> this....<br>><br>> externalIP= x.x.x.x<br>> localnet= x.x.x.<br>><br>> hope this can help you....<br>><br>><br>> On 12/8/05, Moises Silva <<a href="mailto:moises.silva@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
moises.silva@gmail.com</a>><br>> wrote:<br>> ><br>> > what type of NAT do you have? sync? full cone?<br>> cone restricted, port<br>> > restricted?<br>> > any messages in asterisk verbose console?
<br>> ><br>> > best regards<br>> ><br>> > On 12/7/05, chawki hammoud <<a href="mailto:cyhammoud@yahoo.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">cyhammoud@yahoo.com
</a>><br>> wrote:<br>> > ><br>> > > Hi list:
<br>> > > i have an asterisk box behind the NAT ,when i<br>> try to<br>> > > send calls through Sip to the voip provider<br>> server the<br>> > > call is answered but in a one way calling,I hear
<br>> the<br>> > > voice of the other side just for 4 seconds and<br>> then<br>> > > stop but the call do not hangup.<br>> > ><br>> > > my sip.conf is:<br>> > > [voip provider]
<br>> > > type=peer<br>> > > host=<a href="http://213.112.50.8" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">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 href="http://dsl.yahoo.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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<br>> ><br>><br>><br>> --<br>> Jeffery<br>> > _______________________________________________<br>> --Bandwidth and Colocation provided by <a href="http://Easynews.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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