> 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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