search for: natip

...--set-mark 66 iptables -t nat -A POSTROUTING -o eth2 -p tcp --dport 80 -s 192.168.0.0/24 -d ! 192.168.0.0/16 -j MASQUERADE iptables -t nat -A POSTROUTING -o eth2 -p tcp --dport 3128 -s 192.168.0.0/24 -d ! 192.168.0.0/16 -j MASQUERADE i have also /sbin/ip route add 192.168.0.0/24 dev eth1 table natips /sbin/ip route add 127.0.0.0/8 dev lo scope link table natips /sbin/ip route add default via 217.10.248.1 dev eth2 table natips /sbin/ip route flush cache /sbin/ip rule add fwmark 66 table natips squid is running on 192.168.0.1:3128 without squid it''s working i''m using secon...

...have this settings and without squid it''s working i have default route on the first internet connection. iptables -t nat -I POSTROUTING -o eth2 -p tcp --dport 80 -s 192.168.0.0/24 -d ! 192.168.0.0/16 -j SNAT --to 217.10.248.135 /sbin/ip route add default via 217.10.248.135 dev eth2 table natips /sbin/ip rule add fwmark 66 table natips iptables -t mangle -I PREROUTING -i eth1 -p tcp --dport 80 -j MARK --set-mark 66 iptables -t mangle -A FORWARD -i eth1 -p tcp --dport 80 -j MARK --set-mark 66 I try to solve the problem moving squid to onother computer and i add additional rules lik...

...68.0.2 I have : iptables -t mangle -I PREROUTING -i eth1 -s 192.168.0.0/24 -d mail.mail.org -p tcp --dport 110 -j MARK --set-mark 67 iptables -t mangle -I PREROUTING -i eth1 -s 192.168.0.0/24 -d mail.mail.org -p tcp --dport 25 -j MARK --set-mark 67 /sbin/ip route add 192.168.0.0/24 dev eth1 table natips /sbin/ip route add 127.0.0.0/8 dev lo scope link table natips /sbin/ip route add default via 192.168.0.2 dev eth1 table natips /sbin/ip route flush cache /sbin/ip rule add fwmark 67 table mail But it''s not working . root@fw:/usr/src/linux# ip rule list 0: from all lookup local 3...

...ing to an external Asterisk server which is of course not behind NAT. The docs say that nat=yes will cause Asterisk to ignore the IPs in the SIP headers and SDP, and replace them with the actual address the packets are coming from. I thought this meant that if I put any address in the Cisco's NATIP field, this would work because Asterisk would rewrite the SDP to the real address, which might have changed since the NATIP field was set. However, Asterisk is not doing this (though it's doing some other interesting things, now noticing that the NAT address in the Via header is local to it an...

...ng to authenticate (ssl or not) via the webmail interface I get this error... NAT IP is the public ip I assigned. It doesn't work for an AD account or an account local to the Mac server. However it DOES work on the internal LAN. Feb 22 18:15:36 mail dovecot[3501]: auth: Error: od (username,natIP): verify plain: lookup failed for user: username Feb 22 18:15:51 mail dovecot[3501]: auth: Error: od[getpwnam_ext](username,natIP): No record for user I figured I'm missing something in the dovecot config. I have followed apple's kb articles for AD webmail authentication and kerberos, an...

...ister my cisco 7970 remotely to it, but it just won't go. I am running 1.4.2 internally and the phone registers fine to it. I'm using the latest firmware (i think) - 8.2.1S On the server in question I have tried the following for the sip declaration: qualify=never nat=no (yes) defaultip=(natip)(externalip) md5secret=<md5pass> or secret=<secret> Nothing seems to work, and I continually get "sip 401 unauthorized" messages on the console when the phone tries to register. I've spent a number of hours on this googling and searching for anyone working with 1.2 and 7...

Should the following setup work? SIP UA---NAT---Internet---NAT---SIP UA If both UA's support STUN and report the external IP address in the SIP packet.. I am trying to get away from using canreinvite=no so that traffic can go directly between the UA's and not via the central server but I can't seem to get it to work.. Has anyone set this up and can give me some pointers??

Hi! I have this configuration: SIP client A <-> NAT box A (real external IP) <-> Asterisk server (real IP) <-> (real external IP) NAT box B <-> SIP client B The echo test form any of the clients to the asterisk server is working just fine, even without canreinvite=no. When I try to call from SIP client A to B, wihtout the canreinvite=no in the sip.conf, the call

My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k on Debian stable). It could, however, register with another installation of Asterisk and the settings on the phone (apart from the SIP proxy address) haven't changed since then. On the new Asterisk box my sip.conf contains this: [jeremy] type=friend regexten=801 allow=g729 host=dynamic secret=PASSWORD nat=yes

Hi guys, sorry to be iterating this on the list once more, but I'm not able to get this stuff to work as I'd expect. So far, I've always managed to keep it out of NAT environments :-> My home LAN is NATed by a simple Draytek router. In the home LAN is an ATA186 with SIP. On the internet (public) is an Asterisk server. I have nat=yes in the sip.conf and the connectmode is set