[asterisk-users] Searching on how to keep local calls... local

[jonas kellens]

> Your best option without a local asterisk server is to set up the
> remote server to do reinvites when calls are going local->local
> 
> The calls will end up routed through your internet router, but not
> beyond that.


So by placing "canreinvite=yes" in sip.conf, the RTP-traffic would flow
between the 2 IP-phones and through the router.
Do I loose music on hold ? I guess I do...


> Downside: might have to make each ip phone available via port forwards


And if I place "nat=yes" in sip.conf ??
Or will IP-phone 1 not know the local IP-address of IP-phone 2 for
sending a re-invite ??


Jonas.
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