# make TESTIP=boris test
test:
- rcp voiplines.js set_voiplines.pl set_voiplines.sh voiplines.sh voiplines.pl set_ring.sh prototype.js admin.sh index.html faq.sh faq.html about.sh at-530.html ipphones.pl ipphones.sh ipphones.js voipline.jpg anphone.png phoneline.jpg ipphone.png phones.pl tooltips.html phones.sh phones.js tick.png cross.png banner.html menu.html check_loggedin.sh logout.sh login.sh network.sh network.js set_network.sh tooltip.js tooltip.css root@$(TESTIP):/www
- rcp sip.conf extensions.conf root@$(TESTIP):/etc/asterisk
+ rcp cgi-bin/*.html cgi-bin/*.sh cgi-bin/*.pl cgi-bin/*.js cgi-bin/*.css cgi-bin/*.png root@$(TESTIP):/www
+ rcp etc/asterisk/*.conf root@$(TESTIP):/etc/asterisk
--- /dev/null
+Easy Asterisk
+=============
+
+Introduction
+------------
+
+Easy Asterisk is a simple Web GUI for Asterisk with comprehensive tool
+tip documentation and a learning curve of a few minutes. Analog ports
+and IP Phones are automatically detected, and no knowledge of Asterisk
+configuration is required.
+
+Easy Asterisk is "unfeatured" - most of the Asterisk features are
+hidden by the GUI. Instead it tries to make very basic, very common
+Asterisk installations fast and simple, for example:
+
+* An Asterisk server running on your home gateway/firewall/server.
+ You want to connect a few IP Phones and make cheap phones calls
+ using VOIP. An CD based Asterisk distro is a possibility but you
+ don't want to dedicate a full PC just for Asterisk. You don't
+ really want to learn obscure dial plan syntax and yet another conf
+ file format.
+
+* An 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 some phone
+ or computer guy at $100/hr to maintain your phone system.
+
+Implementation
+--------------
+
+Easy Asterisk is written for the
+http://rowetel.com/ucasterisk/ip04.html[IP0X embedded Asterisk]
+hardware but will also run on x86 and probably many other platforms.
+Easy Asterisk does not require a dedicated PC, you do not need to
+reformat a hard disk and install an ISO CD image.
+
+It is implemented as a set of CGI scripts (shell, perl, java-script)
+and pre-configured extensions.conf and sip.conf files. Only a basic
+web server and microperl support are required, making it suitable for
+small embedded platforms. No Perl libraries or CPAN are required.
+
+Easy Asterisk is "unfeatured" - it hides many of the advanced Asterisk
+features in the interest of simple and fast configuration. However
+once you get started and your experience builds these features are of
+course available using regular Asterisk configuration techniques.
+
+Easy Asterisk has been written to be compatible with regular Asterisk
+conf-file style configuration. Just leave the conf file lines with
+"easy-asterisk" comments alone. The Asterisk conf files
+extensions.conf and sip.conf are directly modified by Easy Asterisk,
+but changes are limited to the "easy-asterisk" lines.
+
+No database is used. Neither is the users.conf/AJAX style interface
+based on the Asterisk built-in web server which should be disabled
+when using Easy Asterisk.
+
+For Auto-detection of Zap ports Easy Asterisk looks at
+/etc/zapata.conf, so you may need this set up correctly for your
+analog hardware. On the IP0X this happens automatically.
+
+Installation
+------------
+
+Note this process will overwrite your Asterisk extensions.conf and
+sip.conf files - back them up if you have an existing Asterisk
+installation that you want to keep.
+
+IP0X:
+
+ -------------------------------------------------------------------
+ root~> ipkg install easy-asterisk
+ root~> reboot
+ -------------------------------------------------------------------
+
+x86:
+
+1. 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 /your/web/server/root (lighttpd
+ config instructions below).
+
+2. Login as root to install the Easy Asterisk files:
+
+ -------------------------------------------------------------------
+ # cd ~
+ # svn co https://freetel.svn.sourceforge.net/svnroot/freetel/easy-asterisk-gui
+ # cp easy-asterisk-gui/etc/asterisk/* /etc/asterisk
+ # cp easy-asterisk-gui/cgi-bin/* /your/web/server/root
+ # mv etc/asterisk/users.conf etc/asterisk/users.conf.bak
+ -------------------------------------------------------------------
+
+ The final step above may not be required on your machine if you
+ don't have a users.conf.
+
+3. 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.
+
+4. I use lighttpd as the web server, the /etc/lighttpd.conf lines
+ required are:
+
+ -------------------------------------------------------------------
+ cgi.assign = ( ".sh" => "/bin/sh",".pl" => "/usr/sbin/microperl" )
+ -------------------------------------------------------------------
--- /dev/null
+; extensions.conf
+; David Rowe 4 Jan 2010
+;
+
+; Designed for Easy Asterisk GUI. However you can hand modify as much
+; as you like, as GUI scripts read/and write regular extensions.conf
+; without messing with your custom dialplan
+
+
+[general]
+static = yes
+writeprotect = no
+autofallthrough = yes
+clearglobalvars = no
+priorityjumping = no
+
+[default]
+
+; Pre-configured analog extensions, depends on IP0X model and what modules
+; are installed. Some of these may map to FXO ports
+
+exten => 6001,1,Dial(Zap/1)
+exten => 6002,1,Dial(Zap/2)
+exten => 6003,1,Dial(Zap/3)
+exten => 6004,1,Dial(Zap/4)
+exten => 6005,1,Dial(Zap/5)
+exten => 6006,1,Dial(Zap/6)
+exten => 6007,1,Dial(Zap/7)
+exten => 6008,1,Dial(Zap/8)
+
+; Pre-configured SIP-phone extensions. Primary use case is multiple SIP
+; extensions and FXO analog Ports
+
+exten => 6011,1,Dial(SIP/6011)
+exten => 6012,1,Dial(SIP/6012)
+exten => 6013,1,Dial(SIP/6013)
+exten => 6014,1,Dial(SIP/6014)
+exten => 6015,1,Dial(SIP/6015)
+exten => 6016,1,Dial(SIP/6016)
+exten => 6017,1,Dial(SIP/6017)
+exten => 6018,1,Dial(SIP/6018)
+exten => 6019,1,Dial(SIP/6019)
+exten => 6020,1,Dial(SIP/6020)
+exten => 6021,1,Dial(SIP/6021)
+exten => 6022,1,Dial(SIP/6022)
+exten => 6023,1,Dial(SIP/6023)
+exten => 6024,1,Dial(SIP/6024)
+exten => 6025,1,Dial(SIP/6025)
+exten => 6026,1,Dial(SIP/6026)
+exten => 6027,1,Dial(SIP/6027)
+exten => 6028,1,Dial(SIP/6028)
+exten => 6029,1,Dial(SIP/6029)
+exten => 6030,1,Dial(SIP/6030)
+
+;; Pre-configured outgoing Analog group
+
+exten => _0.,1,Dial(Zap/g1/${EXTEN:1})
+
+;; Pre-configured outgoing VOIP line
+
+exten => _1.,1,Dial(SIP/voip/${EXTEN:1})
+
+;; Pre-configured incoming calls
+
+exten => s,1,Dial(SIP/6011) ;; easy-asterisk - don't remove this comment
+++ /dev/null
-; extensions.conf
-; David Rowe 4 Jan 2010
-;
-
-; Designed for Easy Asterisk GUI. However you can hand modify as much
-; as you like, as GUI scripts read/and write regular extensions.conf
-; without messing with your custom dialplan
-
-
-[general]
-static = yes
-writeprotect = no
-autofallthrough = yes
-clearglobalvars = no
-priorityjumping = no
-
-[default]
-
-; Pre-configured analog extensions, depends on IP0X model and what modules
-; are installed. Some of these may map to FXO ports
-
-exten => 6001,1,Dial(Zap/1)
-exten => 6002,1,Dial(Zap/2)
-exten => 6003,1,Dial(Zap/3)
-exten => 6004,1,Dial(Zap/4)
-exten => 6005,1,Dial(Zap/5)
-exten => 6006,1,Dial(Zap/6)
-exten => 6007,1,Dial(Zap/7)
-exten => 6008,1,Dial(Zap/8)
-
-; Pre-configured SIP-phone extensions. Primary use case is multiple SIP
-; extensions and FXO analog Ports
-
-exten => 6011,1,Dial(SIP/6011)
-exten => 6012,1,Dial(SIP/6012)
-exten => 6013,1,Dial(SIP/6013)
-exten => 6014,1,Dial(SIP/6014)
-exten => 6015,1,Dial(SIP/6015)
-exten => 6016,1,Dial(SIP/6016)
-exten => 6017,1,Dial(SIP/6017)
-exten => 6018,1,Dial(SIP/6018)
-exten => 6019,1,Dial(SIP/6019)
-exten => 6020,1,Dial(SIP/6020)
-exten => 6021,1,Dial(SIP/6021)
-exten => 6022,1,Dial(SIP/6022)
-exten => 6023,1,Dial(SIP/6023)
-exten => 6024,1,Dial(SIP/6024)
-exten => 6025,1,Dial(SIP/6025)
-exten => 6026,1,Dial(SIP/6026)
-exten => 6027,1,Dial(SIP/6027)
-exten => 6028,1,Dial(SIP/6028)
-exten => 6029,1,Dial(SIP/6029)
-exten => 6030,1,Dial(SIP/6030)
-
-;; Pre-configured outgoing Analog group
-
-exten => _0.,1,Dial(Zap/g1/${EXTEN:1})
-
-;; Pre-configured outgoing VOIP line
-
-exten => _1.,1,Dial(SIP/voip/${EXTEN:1})
-
-;; Pre-configured incoming calls
-
-exten => s,1,Dial(SIP/6011) ;; easy-asterisk - don't remove this comment
projects / freetel-svn-tracking.git / commitdiff