asterisk/freeswitch in nat/no-nat setup

Posted by pQd on Server Fault See other posts from Server Fault or by pQd
Published on 2010-06-06T14:36:48Z Indexed on 2010/06/06 14:42 UTC
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hi,

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?

thanks!

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