David Cook wrote:
<snip> 3. Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk <snip>
Then it goes on to say: * #3 Works with port forwarding and some header mangling magic
Can somebody explain a little more about the "header mangling magic" as it is not discussed anywhere else in the document.
No magic required...
Currently I have my firewall port forwarding 5060 to my asterisk server and the UDP port range forwarded as well. Registration works, but no audio. Obviously the RTP stuff is not happy with the forwarding.
In sip.conf ensure:
externip = <external ip> localnet = 255.255.255.0 nat=yes
If it still doesn't work, enable "sip debug" at * CLI to troubleshoot. This will allow you to examine the sip headers to see why the phone & asterisk aren't talking.
Ryan _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
