> > 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... > Try it first, asterisk could just reinvite the audio back to the server Also you might be able to program a SIP address for music on hold into the ip phones
exten => moh,1,Answer() exten => moh,2,MusicOnHold() > > 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 ?? > The remote asterisk server would be doing the reinvites with what it knows > > > Jonas. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
