You can port forward the 5060 SIP port and use externip keyword in sip.conf to have it working behind a NAT.
Martin On Mon, 3 Nov 2003, WipeOut wrote: > Robert Mann wrote: > > > Problem I have is this. outside firewall (extension 2003) can call me > > inside firewall (extension 2000) and all is fine. If I call from > > inside firewall (extension 2000) to outside firewall (extension 2003) > > I hear no ringing and person at other end can pick up and I hear for > > maybe a half second then I go to voicemail. If I add another > > extension on the outside then communication between outside and > > outside through * is not possible at all. I know I can not be the > > only one who has tried to do this. Please any help would be greatly > > appreciated. > > > > Robert, > > You need to get Asterisk onto a public IP address.. Using the DMZ > function on the router will not work.. If you search the archives you > will see that it has been attempted many times.. > > The reason is not in the IP but in the SIP headers.. they will be sent > out from the Asterisk server with the internal IP address of the server, > this means that when the SIP UA reads the SIP message and responds it > will respond to the incorrect IP address.. > > So the basic rules where NAT is involved are.. > > Asterisk server must always be on a public IP address.. > > SIP UA's can be behind NAT but need "nat=yes", "canreinvite=no" and > "qualify=yes" set in the phone configuration in sip.conf.. > > Hope that helps.. > > Later.. > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users