[Asterisk-Users] Call routing based on remote ip address.

[Bruno Hertz]

While setting up my first dial plan, I find that notions like remote
ip, network, or incoming network interface seem to be totally lacking
regarding calling parties, where * still seems to fully rely on the
easily spoofable caller id.

Especially, allowing only certain ips or networks to enter a specific
context in the dial plan is apparently not possible, at least in the
h323 world. Don't know yet about sip or aix, but I guess it's the same
since the extension syntax xyz => extension/somevariable limits
routing decisions to built in variables, where ip related info is
simply missing, at least as far as I can see (you are wholeheartedly
invited to prove me wrong).

Question hence: did anybody tackle those issues anyway, maybe on code
level (patch/extra module)? Are plans underway to fix that stuff? I
just can't believe that, if my above statements were right, anyone
would expose an * server to the internet and still feel secure,
especially if that server allows connections to billable services
(like even bandwidth usually is) ...

Any info highly appreciated.

Thanks, Bruno.