<div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div class="im"><blockquote type="CITE"><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>
<br>
<font size="2">The calls will end up routed through your internet router, but not beyond that.</font><br>
</blockquote>
<br></div>
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...</div></blockquote><div>Try it first, asterisk could just reinvite the audio back to the server <br>Also you might be able to program a SIP address for music on hold into the ip phones<br>
<br>exten => moh,1,Answer()<br>exten => moh,2,MusicOnHold()<br><br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div class="im">
<br>
<br>
<blockquote type="CITE">
<font size="2"><font color="#808080">Downside: might have to make each ip phone available via port forwards</font></font><br>
</blockquote>
<br></div>
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></div></blockquote><div>The remote asterisk server would be doing the reinvites with what it knows <br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div><font color="#888888">
</font><br></div></blockquote><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><font color="#888888">
<br>
Jonas.
</font></div>
<br>_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br>