[Asterisk-Users] howto

[Jörg Bauer/Denic]

I have this configuration:

UA1 ---> FW1 ---> Asterisk ----> FW2 --> Internet  --> UA2

UA has provate address (192.168.x.x)
Asterisk has public address

I want to be reach somebody at the internet.
My idea was that asterisk works as a Proxy.
Then i would have a SIP/RTP connection between UA1 and Asterisk and an 
other SIP/RTP connection between Asterisk and UA2. (asterisk is bridgeing 
these calls). This setup solves the fu....ing NAT problem.

Question 1: 
Is this possible ???

Question 2:
How do I have to configure extension.conf to reach somebody ??

what I want:
my UA1 tries to reach sip:test at example.com 

my config:
exten => .,1,Dial(SIP/${EXTEN})       // whatever you get....try to 
establish a SIP-session ......

but this expands to Dial(SIP/test)    !!!  Domain is gone !!!!


Question 3:
another idea. I also installed a SIP-Proxy. Then the chain is different: 

UA1 ---> FW1(NAT) ---> Asterisk ----> SER (SIP-Proxy) --> FW2 --> Internet 
 --> UA2
I still have the same problem: asterisk always tries  to make a connection 
to test at sip-proxy not to test at example.com


Questions (and Solutions):
A.
am i right that asterisk throws away the domain ??
if yes: 
why don´t we have a environment variable with the Domain ??
example:
exten => .,1,Dial(SIP/${EXTEN}@${DOMAIN})       // whatever you get....try 
to establish a SIP-session ......

B. 
am i right that asterisk does not have the code to use a poxy ??
if yes:
why don´t we create some code to do it ?
example: (sip.conf)

[proxy]
type=proxy
host=10.10.10.10

and
exten => .,1,Dial(SIP/proxy)       // whatever you get....try to establish 
a SIP-session ......

C.
am i right that asterisk doesn´t support RFC3263 ( Session Initiation 
Protocol (SIP): Locating SIP Servers)
if yes:
wouldn´t it be nice to write the code ???

any input is welcome........