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<DIV><SPAN class=039194115-17032003><FONT
face="Book Antiqua">Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=039194115-17032003><FONT face="Book Antiqua">Extensions are
different from H323 endpoints.</FONT></SPAN></DIV>
<DIV><SPAN class=039194115-17032003></SPAN><SPAN class=039194115-17032003><FONT
face="Book Antiqua">My understanding of Cisco routers and Sip/H323 dictates that
there should be a Proxy when routing from 1 Sip/H323 gateway to
another.</FONT></SPAN></DIV>
<DIV><SPAN class=039194115-17032003><FONT face="Book Antiqua">We have tried it
before and the best we ever achieved is a one-way audio in the most minimalist
form.. i.e. Cisco A---->Cisco B------Cisco C.</FONT></SPAN></DIV>
<DIV><SPAN class=039194115-17032003><FONT face="Book Antiqua">In your configs
below, you will be routing to Gateways not endpoints, and that's where the
problem is.</FONT></SPAN></DIV>
<DIV><SPAN class=039194115-17032003><FONT face="Book Antiqua">Perhaps, someone
else has tried this with better luck, but I doubt it.</FONT></SPAN></DIV>
<DIV><SPAN class=039194115-17032003><FONT
face="Book Antiqua">Cheers,</FONT></SPAN></DIV>
<DIV><SPAN class=039194115-17032003><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 3:33 PM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> Re: [Asterisk-Users] Asterisk
as a SIP/H.323 Router<BR><BR></FONT></DIV>
<DIV><FONT face=Arial size=2>Ok, </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> Lets say that I have something like the
following:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> On the Asterisk box I have the following
definitions in sip.conf:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[locationA]<BR>type=friend<BR>host=router.locationA.foobar.com<BR>defaultip=1.1.1.2</FONT></DIV>
<DIV><FONT face=Arial size=2>context=router</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>
<DIV><FONT face=Arial
size=2>[locationB1]<BR>type=friend<BR>host=router.locationB1.foobar.com<BR>defaultip=1.1.2.2</FONT></DIV></DIV>
<DIV>
<DIV><FONT face=Arial size=2>context=router</FONT></DIV></DIV>
<DIV> </DIV>
<DIV>
<DIV><FONT face=Arial
size=2>[locationB2]<BR>type=friend<BR>host=router.locationB2.foobar.com<BR>defaultip=1.1.2.3</FONT></DIV>
<DIV><FONT face=Arial size=2>context=router</FONT></DIV><BR> Now, in
extensions.conf I would have something with the following logic:</DIV>
<DIV> </DIV>
<DIV>[router]<BR>;<BR>; We start with what to do when a call first comes
in.<BR>;<BR>exten => s,1,Wait,1</DIV>
<DIV>exten => s,2,[AGI program that checks the originator and routes
according to pre-defined settings]</DIV>
<DIV> </DIV>
<DIV> Again, this is just a concept, not the actual configuration. But in
theory, I don't a reason why this can't</DIV>
<DIV>work.</DIV>
<DIV> </DIV>
<DIV>Nir Simionovich</DIV>
<DIV> </DIV>
<DIV> </DIV></FONT>
<BLOCKQUOTE dir=ltr
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=alhakeem@softhome.net href="mailto:alhakeem@softhome.net">Abdul
Hakeem</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> Monday, March 17, 2003 4:19
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> RE: [Asterisk-Users] Asterisk as
a SIP/H.323 Router</DIV>
<DIV><BR></DIV>
<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></BLOCKQUOTE></BODY></HTML>