[Asterisk-Users] Asterisk SIP calls failing - not a proxy? What of RTP codec transcoding?

[Ian Blenke]

I have an Asterisk 0.4.0 install working with two grandstream budgetone 
100 phones, gnophone, and kphone. This is a private network segment 
(172.17.x.x), with the PBX configured on my outbound firewall which has 
a public address (66.x.x.x).

- I can make calls between phones - all extensions are working.
- I can make IAX calls to IAXTEL. No problems (apparently gsm only)
- I can call SIP phone numbers
   - The called party can hear me.
   - I cannot hear them.

After looking at the SIP handshaking, it is apparent that Asterisk is 
giving out my Grandstream's private IP address (172.17.x.128) to the 
called party in the INVITE. This is bad.

The documentation suggests that Asterisk terminates all RTP streams and 
does codec transcoding to make negotiated calls to external SIP 
endpoints - very proxy like (the behavior I'm looking for).

How do I configure Asterisk to "hide" origionating SIP phone addresses, 
masquerading as itself instead?

I guess my only option is going to be an Asterisk install on the public 
Internet with *no* private connection and some kind of SIP proxy on my 
firewall firewall (behind which the phones will sit). All phones will 
need to register with the Asterisk PBX through the outbound proxied 
connections.

-- 
- Ian C. Blenke <icblenke at nks.net>
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