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* Cleaning up public/ and documenting js/css libs. This commit mostly addresses #1484 by moving vendor'ed plugins into a vendor/ directory and documenting their upstream source and license in vendor/librejs.html. This also proves gitea is using only open source js/css libraries which helps toward reaching #1524. * Removing unused css file. The version of this file in use is located at: vendor/plugins/highlight/github.css * Cleaned up librejs.html and added javascript header A SafeJS function was added to templates/helper.go to allow keeping comments inside of javascript. A javascript comment was added in the header of templates/base/head.tmpl to mark all non-inline source as free. The librejs.html file was updated to meet the current librejs spec. I have now verified that the librejs plugin detects most of the scripts included in gitea and suspect the non-free detections are the result of a bug in the plugin. I believe this commit is enough to meet the C0.0 requirement of #1534. * Updating SafeJS function per lint suggestion * Added VERSIONS file, per request
155 lines
4.5 KiB
HTML
155 lines
4.5 KiB
HTML
<!doctype html>
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<title>CodeMirror: Asterisk dialplan mode</title>
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<meta charset="utf-8"/>
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<link rel=stylesheet href="../../doc/docs.css">
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<link rel="stylesheet" href="../../lib/codemirror.css">
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<script src="../../lib/codemirror.js"></script>
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<script src="asterisk.js"></script>
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<style>
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.CodeMirror {border: 1px solid #999;}
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.cm-s-default span.cm-arrow { color: red; }
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</style>
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<div id=nav>
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<a href="http://codemirror.net"><h1>CodeMirror</h1><img id=logo src="../../doc/logo.png"></a>
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<ul>
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<li><a href="../../index.html">Home</a>
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<li><a href="../../doc/manual.html">Manual</a>
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<li><a href="https://github.com/codemirror/codemirror">Code</a>
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</ul>
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<ul>
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<li><a href="../index.html">Language modes</a>
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<li><a class=active href="#">Asterisk dialplan</a>
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</ul>
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</div>
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<article>
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<h2>Asterisk dialplan mode</h2>
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<form><textarea id="code" name="code">
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; extensions.conf - the Asterisk dial plan
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;
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[general]
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;
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; If static is set to no, or omitted, then the pbx_config will rewrite
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; this file when extensions are modified. Remember that all comments
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; made in the file will be lost when that happens.
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static=yes
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#include "/etc/asterisk/additional_general.conf
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[iaxprovider]
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switch => IAX2/user:[key]@myserver/mycontext
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[dynamic]
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#exec /usr/bin/dynamic-peers.pl
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[trunkint]
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;
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; International long distance through trunk
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;
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exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
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exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})
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[local]
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;
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; Master context for local, toll-free, and iaxtel calls only
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;
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ignorepat => 9
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include => default
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[demo]
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include => stdexten
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;
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; We start with what to do when a call first comes in.
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;
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exten => s,1,Wait(1) ; Wait a second, just for fun
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same => n,Answer ; Answer the line
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same => n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
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same => n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
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same => n(restart),BackGround(demo-congrats) ; Play a congratulatory message
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same => n(instruct),BackGround(demo-instruct) ; Play some instructions
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same => n,WaitExten ; Wait for an extension to be dialed.
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exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
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exten => 2,n,Goto(s,instruct)
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exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french
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exten => 3,n,Goto(s,restart) ; Start with the congratulations
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exten => 1000,1,Goto(default,s,1)
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;
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; We also create an example user, 1234, who is on the console and has
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; voicemail, etc.
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;
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exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
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; (but skip if channel is not up)
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exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
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exten => 1234,n,Goto(default,s,1) ; exited Voicemail
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exten => 1235,1,Voicemail(1234,u) ; Right to voicemail
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exten => 1236,1,Dial(Console/dsp) ; Ring forever
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exten => 1236,n,Voicemail(1234,b) ; Unless busy
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;
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; # for when they're done with the demo
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;
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exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
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exten => #,n,Hangup ; Hang them up.
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;
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; A timeout and "invalid extension rule"
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;
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exten => t,1,Goto(#,1) ; If they take too long, give up
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exten => i,1,Playback(invalid) ; "That's not valid, try again"
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;
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; Create an extension, 500, for dialing the
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; Asterisk demo.
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;
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exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
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exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo
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exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
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exten => 500,n,Goto(s,6) ; Return to the start over message.
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;
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; Create an extension, 600, for evaluating echo latency.
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;
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exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
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exten => 600,n,Echo ; Do the echo test
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exten => 600,n,Playback(demo-echodone) ; Let them know it's over
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exten => 600,n,Goto(s,6) ; Start over
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;
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; You can use the Macro Page to intercom a individual user
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exten => 76245,1,Macro(page,SIP/Grandstream1)
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; or if your peernames are the same as extensions
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exten => _7XXX,1,Macro(page,SIP/${EXTEN})
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;
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;
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; System Wide Page at extension 7999
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;
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exten => 7999,1,Set(TIMEOUT(absolute)=60)
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exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
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; Give voicemail at extension 8500
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;
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exten => 8500,1,VoicemailMain
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exten => 8500,n,Goto(s,6)
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</textarea></form>
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<script>
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var editor = CodeMirror.fromTextArea(document.getElementById("code"), {
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mode: "text/x-asterisk",
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matchBrackets: true,
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lineNumber: true
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});
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</script>
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<p><strong>MIME types defined:</strong> <code>text/x-asterisk</code>.</p>
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</article>
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