file format.
* A small office that already has an old analog phone system. You
- want to keep your current analog lines for incoming calls, but
- install 8 IP Phones and use VOIP for outgoing calls. You know
- enough to set up a DSL router but don't want to rely on "the Phone
- Guy" or "The Computer Guy" at $100/hr to maintain your phone
- system.
+ want to keep your current analog lines for incoming calls, but use
+ IP Phones for handsets and VOIP for outgoing calls. You know enough
+ to set up a DSL router and don't want to rely on "the Phone Guy" or
+ "The Computer Guy" at $100/hr to maintain your phone system.
* You are a "Phone Guy" who doesn't understand Linux and Asterisk but
you want to install IP-PBXes.
Asterisk GUIs] out there already. So why do we need another one?
Well I needed an Asterisk GUI that was very easy to use for the
-link:ip04.html[IP0X devices] I sell. Something that would lower the
+link:ip04.html[IP0X] family. Something that would lower the
technical skill required to install and maintain an Asterisk Phone
system. Something my wife and kids could use.
* Light weight so it can run on embedded boxes like the IP0X family.
No SQL database or PHP or LAMP. Only a basic web server and a very
- basic perl are required (e.g. microperl - no CPAN libraries).
+ basic perl are required (microperl - no CPAN libraries).
* Works directly on extensions.conf and sip.conf, but honors any edits
you make to these files. So all the powerful Asterisk features are
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
- get a mention.
+ get mentioned.
* Extensive tool tip documentation. No manual.
* Extensive pre-configuration of extensions.conf and sip.conf,
pre-selected phone numbers, SIP trunks are selected from a pull-down
menu (except they are called VOIP lines). Analog ports are auto
- detected, at least on the IP0X. This makes adding IP phones very
- fast and simple.
+ detected, at least on the IP0X. This makes adding phones fast and
+ simple.
[[status]]
Status
Alpha:
* Works on IP0X.
-* Not tested on x86. Several IP0X features are n/a for x86 and should be
- disabled when the x86 (or non-IP0X) platform is detected.
+* Works on x86 but many features (like upgrades) disabled in the
+ interests of security. These need to be implemented in a safe way.
* Needs feedback from real users to see how useful the concept is and what
(un)features need to be added.
* Need a few more (un)features to be added, and Voip Line screen
populated with more ITSPs.
-* But quite useable as it stands.
+* But quite useable as it stands, especially as a starting point for
+ IP0X.
[[notes]]
Implementation Notes
-------------------------------------------------------------------
# cd ~
# svn co https://freetel.svn.sourceforge.net/svnroot/freetel/mini-asterisk-gui
+# cd mini-asterisk-gui
# ./update_revision.sh
-# cp mini-asterisk-gui/etc/asterisk/* /etc/asterisk
-# cp mini-asterisk-gui/cgi-bin/* /www
+# cp etc/asterisk/* /etc/asterisk
+# cp cgi-bin/* /var/www
# cd /etc/asterisk
# cp extensions.conf extensions.conf.def
# cp sip.conf sip.conf.def
-# mv etc/asterisk/users.conf etc/asterisk/users.conf.bak
--------------------------------------------------------------------
+# mv users.conf users.conf.bak
+
-The final step above may not be required on your machine if you don't
-have a users.conf. The .def copies are required by the "reset
-defaults" feature on the admin screen.
+The .def copies are required by the "reset defaults" feature on the
+admin screen.
-. Switch off the internal Asterisk web server by editing
+. It's a good idea to switch off the internal Asterisk web server by editing
/etc/asterisk.httpd.conf. Make sure the enabled line reads like this:
enabled=no
+
Then stop and restart Asterisk.
++
+-------------------------------------------------------------------
+# /etc/init.d/asterisk restart
+-------------------------------------------------------------------
++
+. I use lighttpd as the web server. On my x86 box I needed to install
+a sym-link and edit the 10-cgi.conf file:
++
+-------------------------------------------------------------------
+# cd /etc/lighttpd/conf-enabled
+# ln -s ../conf-available/10-cgi.conf .
+-------------------------------------------------------------------
++
+I then modified 10-cgi-conf to enable perl and shell scripts:
++
+-------------------------------------------------------------------
+cgi.assign = (
+ ".pl" => "/usr/bin/perl",
+ ".sh" => "/bin/sh",
+)
+-------------------------------------------------------------------
++
+Make sure lighttpd runs as root, as we need to run Asterisk etc from
+shell script CGIs:
++
+## change uid to <uid> (default: don't care)
+#server.username = "wwwrun"
-. I use lighttpd as the web server, the /etc/lighttpd.conf lines
-required are:
+## change uid to <uid> (default: don't care)
+#server.groupname = "wwwrun"
++
+Restart lighttpd after all the changes:
++
+--------------------------------------------------------------------
+# /etc/init.d/lighttpd restart
+------------------------------------------------------------------
++
+For comparison on the IP04 I just modified /etc/lighttpd.conf:
+
-------------------------------------------------------------------
cgi.assign = ( ".sh" => "/bin/sh",".pl" => "/usr/bin/perl" )
Contributions
-------------
-I welcome sip.conf entries for your favourite ITSP (VOIP service) to
-help populate the Provider field of the
+I especially welcome sip.conf entries for your favourite ITSP (VOIP
+service) to help populate the Provider field of the
link:mini/voiplines.sh.html[Voip Line Screen].
+Suggestions and patches for new features are also very welcome.
+
[[support]]
Support
-------
Comments, features request, bugs please let me know using Free
-Telephony Project http://www.rowetel.com/ucasterisk/#support[Support].
+Telephony Project
+https://lists.sourceforge.net/lists/listinfo/freetel-discuss[Mailing
+List]
[[source]]
Source Code
setting up the CGI QUERY_STRING environment variable:
[david@host cgi-bin]$ export QUERY_STRING="pass=uClinux" ; sh login.sh
+
+. When editing on an x86 host and testing on an IP0X Emacs has a nice
+ feature M-! that lets you execute a shell command. I use this to
+ download the shell fuile I am working on, e.g. "scp admin.sh
+ root@ip04".
# David Rowe 7 Jan 2010
# Admin screen for Mini Asterisk GUI
-# check we are logged in
+# check we are logged in ----------------------------------------
echo $HTTP_COOKIE | grep "loggedin" > /dev/null
if [ $? -eq 1 ]; then
exit
fi
+# check what sort of machine we are running on. Many operations
+# outlawed for x86
+
+cat /proc/cpuinfo | grep "CPU:.*ADSP" > /dev/null
+if [ $? -eq 0 ]; then
+ mach="ip0x"
+else
+ mach="x86"
+fi
+
# set password CGI -----------------------------------------------
echo "$QUERY_STRING" | grep -oe "pass=" > /dev/null
if [ $? -eq 0 ]; then
+ if [ $mach != "ip0x" ] ; then
+ echo "<html>Sorry, only supported on the IP0X</html>"
+ exit
+ fi
+
pass=`echo "$QUERY_STRING" | grep -oe "pass=[^&?]*" | sed -n "s/pass=//p"`
passwd_cmdline $pass
fi
echo "$QUERY_STRING" | grep -oe "restart=1" > /dev/null
if [ $? -eq 0 ]; then
+ if [ $mach != "ip0x" ] ; then
+ echo "<html>Sorry, only supported on the IP0X</html>"
+ exit
+ fi
+
# kill cookie to log out. This ensures hitting refresh wont run
# the restart process again
asterisk -rx "dialplan reload" 2>/dev/null 1 > /dev/null
fi
-# Upgrade Mini Asterisk CGI ----------------------------------------------------
+# Upgrade Mini Asterisk CGI -------------------------------------------
echo "$QUERY_STRING" | grep -oe "upgrade=1" > /dev/null
if [ $? -eq 0 ]; then
# test if we are running on an IP0X
which ipkg >> /dev/null
if [ $? -eq 1 ]; then
- cat <<EOF
- <html>
- <head>
- <title>Mini Asterisk - Upgrade</title>
- </head>
- <body>
- Sorry upgrade only supported on IP0X at this time
- </body>
- </head>
- </html>
-EOF
+ echo "<html>Sorry, only supported on the IP0X</html>"
+ exit
fi
cat <<EOF
echo "$QUERY_STRING" | grep -oe "firmwareurl=" > /dev/null
if [ $? -eq 0 ]; then
- h=`hostname`
- if [ ! $h == "ip04" ]; then
- echo "<html>Sorry Firmware Upgreade only supported on IP04</html>"
+ if [ $mach != "ip0x" ] ; then
+ echo "<html>Sorry, only supported on IP0X</html>"
+ exit
fi
firmwareurl=`echo "$QUERY_STRING" | grep -oe "firmwareurl=[^&?]*" | sed -n "s/firmwareurl=//p"`
<ol>
-<li><strong>Transfer a Call</strong>
+<li><strong>Where can I get Support:</strong> You can try the
+ <a href="https://lists.sourceforge.net/lists/listinfo/freetel-discuss">Free Telephony Project</a> Mailing List.
-<li><strong>Answer a call on another ringing phone</strong>: If you hear another phone ringing and
-want to answer the call on your phone, just pick up your phone and
-dial *8. Note an IP phone can pick up a call on a ringing IP phone
-but not an Analog phone. An Analog pohone can pick up another
-ringing Analog phone but not an IP phone.
+<li><strong>Answer a call on another ringing phone</strong>: If you
+ hear another phone ringing and want to answer the call on your
+ phone, just pick up your phone and dial *8. Note an IP phone can
+ pick up a call on a ringing IP phone but not an Analog phone. An
+ Analog pohone can pick up another ringing Analog phone but not an
+ IP phone.
+
+<li><strong>Put a call on hold/transfer:</strong> On an IP Phone try
+ the recall or transfer button. On an Analog Phone hold down the
+ hook button for 1 second then dial the new number.
+
+<li><strong>Why another Asterisk GUI</strong> I had some specific
+ needs for a light weight, very easy to use GUI for
+ the <a href="http://www.rowetel.com/ucasterisk/ip04.html">IP0X</a>
+ line of products. So I scratched the itch. More
+ information <a href="http://www.rowetel.com/ucasterisk/mini.html#why">here</a>.
</ol>
projects / freetel-svn-tracking.git / commitdiff