my current setup – i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured – one on internal sip proxy [for calls between the branches], another – at 3rd party voip providers [ since it's in different countries – those are different providers, but that's irrelevant ].
i was thinking about terminating sip calls on something like asterisk/freeswitch server and having all sip-devices log on just once to such server[s] – mostly to provide things like voicemail, groupcalls, redirections etc. it seems perfectly doable but there is one problem – i cannot find examples how to prepare for nat/no nat. for calls routed to from/to 3rd party voip operator – i'll need handling for nat/stun etc, but for handling of internal calls – i do not want any nat, all traffic should go via vpns to different branches.
can you provide me some hints how to configure it? any tutorials?