We looked at this before going to * as our main system at the corporate headquarters. We have not rolled * out to our other sites yet, but in our case each site has local phone lines and wan contectivity so we are planing a full pbx at each site and then trunking. In theory you could dial the remote site via your prefered trunk, like IAX and if you detected a channel unavaliable message have a backup route in the dial plan like a DID into the system somewhere. Like I said we did not plan on using the main * servers at corporate to handle all calling.
<br><br><br><div><span class="gmail_quote">On 6/26/06, <b class="gmail_sendername">Curt Shaffer</b> <<a href="mailto:cshaffer@gmail.com">cshaffer@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div link="blue" vlink="purple" lang="EN-US">
<div>
<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">Has anyone out there figured out how to emulate the Cisco
SRST functionality with *? If so would you mind letting me know the best
practices for this?</span></font></p>
<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;"> <br>Thanks</span></font></p>
<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">Curt</span></font></p>
</div>
</div>
</div>_______________________________________________<br>--Bandwidth and Colocation provided by <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://Easynews.com" target="_blank">Easynews.com</a> --<br>
<br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users
</a><br><br><br></blockquote></div><br><br clear="all"><br>-- <br>Bruce<br>Nortex Networks