[Users] openser setup as a telephony gateway (to PSTN) for NAT'd clients

I may be misunderstanding things (very probably so), but all the examples for both SER and OpenSER that I have seen either do not do NAT or NAT but are not the "end point" for a UAC to the PSTN. From what I can tell they are just sending on the SIP request to the next destination as is. The setup I am trying to accomplish is * SIP clients are all on a private network with connectivity to OpenSER directly on one interface (on the same private network) * OpenSER's other interface is on the external network (internet facing) * SIP clients are only sending telephone numbers (sip:telephone_number@*... do not know about PSTN providers) * OpenSER connects to the PSTN provider and send the number to dial that came from the SIP client * OpenSER "proxies" the entire call (with the help of rtpproxy or mediaproxy for RTP of course) * No incoming calls from the internet to OpenSER (no support for that is needed) * Registration not required for sip clients (they are all on same private network and authorized) I have found several posts, example configs, documents that have pieces of what I need (from what I can tell).. and I have tried to put it together, but it does not quite work... So is there some example that fits this type of usage? If not one then possibly several pieces from a few documents? I have been thinking that OpenSER setup as a outbound proxy configured for multihome, but everything I have seen on that just routes the calls through to where ever the SIP client was requesting as a final destination and I need to send to one destination (the PSTN provider). Any help is greatly appreciated. BTW, I was thinking the NAThelper or outbound proxy example from http://www.voip-info.org/wiki-SER+tips+and+tricks Looked close to what is needed, but have a bunch of stuff (seemingly) unneeded for my scenario and have nothing about PSTN connectivity. Thanks for taking the time to read this long post (if you made it this far). Taylor