asterisk users - Nov 2004 - SIP clients <--> SE R <--> Asterisk <--> carrier/gateway

Hello,

I'm currently trying to setup a SIP environment for VoIP calling for my
final school project, so I'm just working with VoIP/SIP for 2 weeks.

I'm using SER as a SIP proxy server, but the carrier/gateway I am using for
calling to/from PSTN is requiring me to register at their server and
authorize outgoing calls, which is something SER won't do. So I got the idea
to use asterisk between the PSTN carrier and SER for the authorization,
since Asterisk can register and auth itself.


SIP  ---------   SIP     ---------    SIP    ---------   PSTN
-----|       |-----------|       |-----------|       |------
     ---------           ---------           ---------
       SER               Asterisk             Carrier
					<-- auth stuff -->
	  <--  sip relay  -->

So all asterisk needs to do is register itself at the carrier (I've got that
to work with sip.conf) and relay the incoming calls to ser for further
routing, I got that to work a bit.

I've setup 2 sip extensions in extensions.conf in the default context
(asterisk itself is listening at 5065)

	[globals]
	SERADDRESS=myserbox:5060
	CARRIER=carrier:5060

	[default]
	exten => sip_incomming_from_carrier, 1,
Dial(SIP/{EXTEN}@${SERADDRESS},20,r)
	exten => sip_incomming_from_ser, 1,
Dial(SIP/${EXTEN}@${CARRIER},20,r)

When I get an incoming call from the carrier it gets routed to the SER
server (wrong SIP uri however, it now gets
sip_incomming_from_carrier@myserbox, but that's due to the {EXTEN}, that
should be called_numer@myserbox), but replies from my SER box are not
getting back to the carrier, so if a user is not found (SIP/2.0 404) I see
that message on the Asterisk console (Got SIP response 404 "Not Found"
back
from myserbox), but it isn't relayed to the carrier.

I'm also talking with the carrier about skipping the authorization (or
moving it to a lower layer IE vpn oid), but I like to have a solution ready
if the carrier doesn't want that.

Kind regards,

E. Versaevel