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<DIV><SPAN class=151091614-17032003><FONT
face="Book Antiqua">Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=151091614-17032003><FONT face="Book Antiqua">I took a look at
the architecture.</FONT></SPAN></DIV>
<DIV><SPAN class=151091614-17032003><FONT face="Book Antiqua">The way the Cisco
boxes will work if Asterisk in the middle is a Proxy, but it's
not.</FONT></SPAN></DIV>
<DIV><SPAN class=151091614-17032003><FONT face="Book Antiqua">You cannot
re-direct an incoming Voip call from one gateway to another, only a proxy can do
that.</FONT></SPAN></DIV>
<DIV><SPAN class=151091614-17032003><FONT face="Book Antiqua">The Asterisk in
this mode can only terminate the calls via the PSTN. If it attempts to re-direct
the call to the Cisco, it has to be via it's PRI interface (i.e. Cisco PRI0-3 is
connected to the PRI interface of the Asterisk Location B.</FONT></SPAN></DIV>
<DIV><SPAN class=151091614-17032003><FONT
face="Book Antiqua">Cheers,</FONT></SPAN></DIV>
<DIV><SPAN class=151091614-17032003><FONT
face="Book Antiqua">Abdul</FONT></SPAN></DIV>
<DIV></DIV>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B>
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] <B>On Behalf Of </B>Nir
Simionovich<BR><B>Sent:</B> Monday, March 17, 2003 10:08 AM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> [Asterisk-Users] Asterisk as
a SIP/H.323 Router<BR><BR></FONT></DIV>
<DIV><FONT face=Terminal size=2>Hi All,<BR><BR> I've been spending the
last month experimenting with Asterisk, and I must say that all results
point<BR>to a very positive outcome. <BR><BR> Now, i've been asked the
following question: Is it possible to put an Asterisk box between 2
Cisco<BR>routers or other SIP complianet equiment, then routing SIP/H.323 calls
between the two routers?</FONT></DIV>
<DIV><FONT face=Terminal size=2></FONT> </DIV>
<DIV><FONT face=Terminal size=2> Here's a drawing that will
explain:</FONT></DIV>
<DIV><FONT face=Terminal size=2></FONT> </DIV>
<DIV><FONT face=Terminal size=2>
+--------------+
+----------+
+--------------+<BR> PRI
|
|1.1.1.2
1.1.1.3|
|1.1.2.1
1.1.2.2|
|<BR> ---->+ Cisco Router +-------------------+ Asterisk
+--------+--------+ Cisco Router +-->
PRI0-3<BR> | Location A
|
|Location B|
|
|
|<BR>
+--------------+
+----------+
|
+--------------+<BR>
|<BR>
|
+--------------+<BR>
|
1.1.2.3|
|<BR>
+--------+ Cisco Router +-->
PRI4-7<BR>
|
|<BR>
+--------------+</FONT></DIV>
<DIV><FONT face=Terminal size=2></FONT> </DIV>
<DIV><FONT face=Terminal size=2>The question here is this:</FONT></DIV>
<DIV><FONT face=Terminal size=2></FONT> </DIV>
<DIV><FONT face=Terminal size=2>A phone call is is made to the router in
location A.<BR>The Cisco router routes the call via SIP to the Asterisk Box at
Location B.<BR>According to a set of rules on the Asterisk box, the box would
route the SIP/H.323<BR>to one of the the other Cisco boxes, in order to
terminate the call.</FONT></DIV>
<DIV><FONT face=Terminal size=2></FONT> </DIV>
<DIV><FONT face=Terminal size=2>What do you think, is this
possible?</FONT></DIV>
<DIV><FONT face=Terminal size=2></FONT> </DIV>
<DIV><FONT face=Terminal size=2>Nir Simionovich</FONT></DIV>
<DIV><FONT face=Terminal size=2></FONT> </DIV>
<DIV><FONT face=Terminal size=2>P.S.</FONT></DIV>
<DIV><FONT face=Terminal size=2> Excuse me for the poor ASCII art, I
didn't want to attach a Visio or a PDF file.</FONT></DIV></BODY></HTML>