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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>You might look at installing FXS adapters
at the remote sites which would take a call from the C.O. and then pass it to
the Asterisk system at the Main site. Then you could either use the SIP phones
or IAXy adapters at the remote sites.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>This would in essence terminate all the
lines for all offices in the Main office and then you can write the dial plan
accordingly.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Tim.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Jim Dossey<br>
<b><span style='font-weight:bold'>Sent:</span></b> Monday, November 15, 2004
3:58 PM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> [Asterisk-Users] Question
about remote POTS lines</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>I have a client who asked me about a situation they have. They
have a main office and 3 remote offices. We are installing an Asterisk
server at the main office with SIP phones in the remotes. Each remote
office only has 1 person. The remote offices currently have a POTS line
that has a published number. They want to keep that number. The
problem is that they would like to somehow link those remote POTS lines back to
the main office, so people in the main office can answer their calls when they
are away. They could install an asterisk server in those remote offices
and link them back to the main office, but that seems like overkill for a
single POTS line.<br>
<br>
Any ideas? <o:p></o:p></span></font></p>
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