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<DIV><FONT face=Arial size=2>Amir,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><A
href="http://www.voip-info.org">http://www.voip-info.org</A></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>;)</FONT></DIV>
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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=amir1972@yahoo.com href="mailto:amir1972@yahoo.com">Amir Aziz</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, November 28, 2005 1:31
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] beginner
questions </DIV>
<DIV><BR></DIV>
<DIV>Dear List Members,</DIV>
<DIV> </DIV>
<DIV>I am trying to setup a small asterisk box. My configure is pretty basic
for now. my <SPAN class=docEmphasis>zaptel.conf</SPAN> is as follows </DIV>
<DIV> </DIV>
<DIV> fxoks=1<BR><STRONG><EM>
</EM></STRONG>fxsks=2<BR> loadzone=us<BR>
defaultzone=us</DIV>
<DIV> </DIV>
<DIV>when I run <STRONG># /sbin/ztcfg -vv </STRONG>I get<BR></DIV>
<DIV>Zaptel Configuration<BR>
======================<BR><BR><BR> Channel
map:<BR><BR> Channel 01: FXO Kewlstart (Default) (Slaves:
01)<BR> Channel 02: FXS Kewlstart (Default) (Slaves:
02)<BR><BR> 2 channels configured.</DIV>
<DIV> </DIV>
<DIV>my <SPAN class=docEmphasis>zapata.conf is as follows</SPAN></DIV>
<DIV><SPAN class=docEmphasis></SPAN> </DIV>
<DIV><SPAN class=docEmphasis></SPAN><SPAN class=docEmphasis>[trunkgroups]<BR>;
define any trunk groups<BR><BR>[channels]<BR>; hardware channels<BR>;
default<BR>usecallerid=yes<BR>hidecallerid=no<BR>callwaiting=no<BR>threewaycalling=yes<BR>transfer=yes<BR>echocancel=yes<BR>echotraining=yes<BR>immediate=no<BR><BR>;
define channels<BR><SPAN
class=docEmphStrong>context=internal
; Uses the [internal] context in
extensions.conf<BR>signalling=fxo_ks ; Use
FXO signalling for an FXS channel<A name=IDX-CHP-4-0902></A> <BR>channel
=> 1 ;
Telephone attached to port
1</SPAN><BR><BR>context=incoming ;
Incoming calls go to [incoming] in
extensions.conf<BR>signalling=fxs_ks ; Use
FXS signalling for an FXO channel<BR>channel =>
2 ; PSTN at
tached to port 2</SPAN></DIV>
<DIV><SPAN class=docEmphasis></SPAN> </DIV>
<DIV>I have also configured two soft phones (xten). I can dial from one
softphone to the other and vice versa. I am using SIP. I am using O'Reilly
Asterisk: The Future of Telephony. Now to the questions I hope I have provided
enough information I am using the latest release of Redhat Linux and Asterisk.
I have got 1 FXS and 3 FXO channels.</DIV>
<DIV> </DIV>
<DIV>1. I do not get any kind of tone in my analogie phone connected to
the FXS port.</DIV>
<DIV>2. when I run ZTTOOL I see the card but are there any other tools that
give more information I do not see any activity going on.</DIV>
<DIV>3. when I turn on computer all the lights at the back of digium card are
green and are list even though no wire is inserted. Is that normal?</DIV>
<DIV>4. when I insert analogue phone line from telco in FXO port what am I
suppose to look at in the zttool. I do not see any kind of activity so
far.</DI V>
<DIV>5. Are there any open source graphical tools that I can also instal to
configure, monitor and troubleshoot asterisk.</DIV>
<DIV>6. What other books/links can be helpful in learning this interesting
software.</DIV>
<DIV> </DIV>
<DIV>I thank you all for you help in advance.</DIV>
<DIV> </DIV>
<DIV>Regards,</DIV>
<DIV>Amir Aziz<BR></DIV>
<DIV> </DIV>
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