[Asterisk-Users] * with SIP behind NAT

[Tjardick van der Kraan]

Hello Everyone,

My * is behind a NAT connection, for which the IAX protocol works perfectly
btw, but i was looking into getting sip to work. I forwarded the 5060 port
to * which seems to work for connecting and outgoing data streams, but
offcourse the incomming voice isn't working.

Is there a way to supply * wich a port range and ip/hostname of the external
host to be used for SIP ?

so for example:

rtp_port_start=8766
rtp_IP=xxx.xxx.xxx.xxx

and maybe even a range for which not to use these ports

rtp_local=192.168.0.0/24

Just some ideas, maybe some of them are allready implemented or on the todo
list but i couldn't find anything about it on the mailinglist archive
myself.

Greetings,

Tjardick