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<DIV><FONT face=Arial size=2>First, you have to configure your zapata.conf &
sip.conf to support your hardware (see <A
href="http://www.voip-info.org/wiki/index.php?page=Asterisk">http://www.voip-info.org/wiki/index.php?page=Asterisk</A> and
read </FONT></DIV>
<DIV><FONT face=Arial size=2><A
href="http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip">http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip</A> ,this
last is a must-read one)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>After that, you have to see if all incoming calls
are done within work hours:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[incoming]</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>exten =>
s,1,GotoIfTime(9:00-15:00,mon-fri,*,*?accepted,1,10)</FONT></DIV>
<DIV><FONT face=Arial size=2>exten =>
s,2,thefunctionyouwantdoneifisnthourtime</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[accepted]</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>exten =>
1,1,Dial(thephoneyouwantasmaster)</FONT></DIV>
<DIV><FONT face=Arial size=2>; add here what you want if that phone is in
use </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>as you see, you have no need for new hardware and
programming is really simple ;)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=jdfalk@internet2.edu href="mailto:jdfalk@internet2.edu">Johnathan
Falk</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=Asterisk-Users@lists.digium.com
href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, November 22, 2005 4:22
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] Master
Telephone</DIV>
<DIV><BR></DIV>
<DIV class=Section1>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">I am the network administrator for
a small school in <st1:place w:st="on"><st1:State
w:st="on">Michigan</st1:State></st1:place>. We are currently using an
older proprietary pbx system and are trying very hard to get away from this
one vender lock in. I have set up an asterisk server using the version 1.2 of
asterisk. Our current system has a master telephone used by the head
secretary that can transfer anyone’s calls and just generally handle all phone
redirection. Kind of like a head receptionist. The one rule we
have about our telephone system is that during school hours a person must
answer. Our superintendent refuses to have a machine answer if there are
people working. So the head secretary must either redirect the calls to
someone who is there or take the call herself. How can we accomplish
this with asterisk? Also if it can’t be accomplished through programming what
special hardware would we be required to
purchase.<o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Johnathan Falk<BR>Network
Administrator</SPAN></FONT><o:p></o:p></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Clinton Community
Schools</SPAN></FONT><o:p></o:p></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">1-517-442-9622</SPAN></FONT><o:p></o:p></P></DIV>
<P>
<HR>
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