doc changes

git-svn-id: https://svn.code.sf.net/p/freetel/code@140 01035d8c-6547-0410-b346-abe4f91aad63

diff --git a/mini-asterisk-gui/README.txt b/mini-asterisk-gui/README.txt
index a4f65226148ff3bbebf768999de2bbb23b3a45b9..e9f07835525ad679a88120466756c189e96658f7 100644 (file)
--- a/mini-asterisk-gui/README.txt
+++ b/mini-asterisk-gui/README.txt
@@ -72,7 +72,7 @@ view!
   magic.  Plain old HTML, a little Java-script and CGIs written in
   shell script and Perl.
 
-* Doesn't use the Asterisk programming "model".  For example you don't
+* Doesn't use the Asterisk programming model.  For example you don't
   have to understand what a dial-plan is, much less understand how to
   code one.  Plain English terms are used instead, for example "Phones
   and Phone lines".  Terms like Asterisk, Linux, SIP, Zap/1 don't even
@@ -159,14 +159,13 @@ x86 Installation
 
 . You need a web server, Asterisk and some sort of Perl installed
 (very basic Perl installation is fine).  Configure your web server to
-run CGIs (.sh and .pl) from /www (lighttpd config instructions below).
+run CGIs (.sh and .pl) from your server root (lighttpd config instructions below).
 Mini Asterisk expects all files (shell, perl, html etc) to be in the
 same directory.  If you find this painful please <<support, let me know>>.
 
-. The process below places the web files in /www, you may like to
-place the files somewhere else like /www/asterisk.  One of the files
-is named index.html so make sure you don't overwrite an existing
-index.html.
+. The process below places the web files in /var/www, you may like to
+place the files somewhere else.  One of the files is named index.html
+so make sure you don't overwrite an existing index.html.
 
 . Login as root to install the Mini Asterisk files:
 +

diff --git a/mini-asterisk-gui/cgi-bin/set_voiplines.pl b/mini-asterisk-gui/cgi-bin/set_voiplines.pl
index 114b7b067780c5f2b79ea3dffdaa840c1ecb6d4f..980931f2e751a3ae980f8edca8f7f5517203d4fd 100644 (file)
--- a/mini-asterisk-gui/cgi-bin/set_voiplines.pl
+++ b/mini-asterisk-gui/cgi-bin/set_voiplines.pl
@@ -45,6 +45,10 @@ while (<SIP>) {
 
            #print "XX provider_new='$provider_new' '$1' '$2' '$3'\n";
                    
+            # note fine /$user seems to be necessary so that incoming VOIP
+           # calls work.  I really doin't get the syntax but messed around for
+           # hours until it worked!
+
            print "register => $user\@$user/$user; $3\n";
        }
        else {

diff --git a/mini-asterisk-gui/cgi-bin/voiplines.sh b/mini-asterisk-gui/cgi-bin/voiplines.sh
index c38295b1b0c77fe7ecb7f43c319eb3bab832cf7a..c7240ed61086857eda4a25655de17d05e0a89344 100644 (file)
--- a/mini-asterisk-gui/cgi-bin/voiplines.sh
+++ b/mini-asterisk-gui/cgi-bin/voiplines.sh
@@ -56,6 +56,8 @@ EOF
 perl ./voiplines.pl
 
 cat <<EOF
+      <tr><td>&nbsp</td></tr>
+      <tr><td colspan="2">Refresh your browser to update Voip Line status </td></tr>
       <tr><td>&nbsp</td></tr>
       <tr><td><input type="submit" value="OK"></td></tr>