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<FONT SIZE="2">Your best option without a local asterisk server is to set up the remote server to do reinvites when calls are going local->local</FONT><BR>
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<FONT SIZE="2">The calls will end up routed through your internet router, but not beyond that.</FONT><BR>
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So by placing "canreinvite=yes" in sip.conf, the RTP-traffic would flow between the 2 IP-phones and through the router.<BR>
Do I loose music on hold ? I guess I do...<BR>
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<FONT SIZE="2"><FONT COLOR="#808080">Downside: might have to make each ip phone available via port forwards</FONT></FONT><BR>
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And if I place "nat=yes" in sip.conf ??<BR>
Or will IP-phone 1 not know the local IP-address of IP-phone 2 for sending a re-invite ??<BR>
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Jonas.
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