First off Isn't RTP a TCP protocol? or am I over tierd again?
Secondly - unless several conditions are met (canreinvite=yes being one of them) it (asterisk) will still proxy the connection. - Check your dial statement for T's ie T and t - the wiki has a full list. David On 4/26/05, Ian Pattison <[EMAIL PROTECTED]> wrote: > Hi All, > > Can anyone help me out here? I'm having some issues configuring my IPTables > firewall to properly NAT SIP and RTP packets to my asterisk server hiding > behind it. > > Here are my current rules: > > #Inbound SIP to HERMES > $IPTABLES -A PREROUTING -t nat -i $EXTIF -p udp --dport 5060 -j DNAT --to > 192.168.123.4:5060 > $IPTABLES -A FORWARD -i $EXTIF -p udp -d 192.168.123.4 --dport 5060 -j ACCEPT > > #Inbound RTP to HERMES > $IPTABLES -A PREROUTING -t nat -i $EXTIF -p udp --dport 10000:20000 -j DNAT > --to 192.168.123.4:10000:20000 > $IPTABLES -A FORWARD -i $EXTIF -p udp -d 192.168.123.4 --dport 10000:20000 -j > ACCEPT > > When I dial out via my SIP provider I appear to get a partial connection (the > phone rings... that's a good sign) but no audio. Inbound I just get a busy > and asterisk sees nothing. SIP SHOW REGISTRY shows me as registered with the > remote host. Something else that worries me is that I'm seeing the good old > "Attempting native bridge..." message when the destination picks up which, to > my understanding, shouldn't happen since I have "canreinvite=no" set for both > my SIP phone and SIP provider. > > Make sense to anyone? > > Ian > > _______________________________________________ > 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 > _______________________________________________ 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
