Have a check through:<br><br><a href="http://www.voip-info.org/wiki-NAT+and+VOIP">http://www.voip-info.org/wiki-NAT+and+VOIP</a><br><br><a href="http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions">http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
</a><br><br>Regards<br>Rob<br><br><div><span class="gmail_quote">On 03/09/06, <b class="gmail_sendername">Siqhamo Sifo</b> <<a href="mailto:siqhamo@newlunar.co.za">siqhamo@newlunar.co.za</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;">
I have ser sitting on my iptables nat box and my asterisk box on the lan<br>. Ser does forwarding so that any requests (register,invite,ack,...) to<br>the nat box at 5060 r sent to my asterisk box on the lan .I can register
<br>from outside<br>to my asterisk box but there is only one way audio , reason being that<br>when the asterisk box sends a sip packet whith session description the sdp<br>part of the sip packet is not natted .I have tried the following :
<br><br> if(src_ip==<a href="http://10.0.0.0/255.0.0.0){">10.0.0.0/255.0.0.0){</a><br> force_rtp_proxy();<br> encode_contact("enc_prefix","wanip");<br> sdp_mangle_ip("
<a href="http://10.0.0.0/255.0.0.0">10.0.0.0/255.0.0.0</a>","wanip");<br> };<br><br><br>and it does not work because my ethernet dump shows that the contact in<br>sdp is<br>
not mangled.<br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:
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