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<DIV><FONT size=2>Channel Associated Signaling is a simple 4 bits 'protocol'
often used in digital non-isdn E1 connections to define/indicate the state of
the line (or channel): sieze, sieze ack, idle, etc, at a given moment.
Usualy, only 2 of those 4 bits are used.</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>However, given that CAS tables may vary from manufacturer to
manufacturer, the most important thing to start with in a PBX-to-PBX CAS
connection is to identify the CAS table being used at the far-end
switch.</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>If you can not easily obtain the tables on the
far-end --which is the usually the case when the brand of
your switch is not the same as the one at the far end, or if you
don't have previous experience with CAS, you may get stuck even if you have a
CAS analyzer.</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>Given that in PBX-to-PBX scenarios you usually go with
E&M signaling, you will also have to deal with the start arrangement;
usually WiNK or IMMediate You have to have the same
start arrangemnet for each direction (REC/XMT) as the far end
switch. If Wink Start, you also have to deal with the length of
the wink pulse.</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>If you have the far end CAS table, just program yours to
match their values.</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>I am new with Asterisk and have never used
zaptel.conf, but if u attach the far end CAS table to your post, we can
be more precise in helping.</FONT></DIV>
<DIV><FONT size=2>Rgds</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>Sam\\\</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV>----- Original Message ----- </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=makrov_k@hotmail.com href="mailto:makrov_k@hotmail.com">M.A.
Ali</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> Wednesday, January 21, 2004 1:05
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] CAS SF Inband
tone signalling problem</DIV>
<DIV><BR></DIV>
<DIV>
<P>Hi,</P>
<P>I am having some problem in defining CAS SF Inband signalling on a Digium
E100P card. The problem is that the syntax given in the sample zaptel.conf
doesn't work. Can someone provide me with an example of the syntax.</P>
<P>I am trying to connnect asterisk with another exchange using a E1 CAS. I
guess SF is the only option as the rest fxo,fxs,e&m etc are all
subscriber end signalling. i need a CO to CO (or PBX to
PBX) signalling...Somebody guide me on that too.</P>
<P>Any help will be highly appreciated.</P>
<P>Thnaks </P>
<P>Sincerly,</P>
<P>Janjua<BR><BR><BR></P>
<DIV> </DIV></DIV><BR clear=all>
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