New plugin audioconf for audio conferences using XMPP (Jitsi) or SIP …

…(Asterisk/FreeSWITCH)

audioconf/audioconf.js

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+(function (root, factory) {
+    if (typeof define === 'function' && define.amd) {
+        define(["converse"], factory);
+    } else {
+        factory(converse);
+    }
+}(this, function (converse) {
+    var _converse = null;
+
+    converse.plugins.add("audioconf", {
+        'dependencies': [],
+
+        'initialize': function () {
+            _converse = this._converse;
+
+            _converse.api.settings.update({
+                visible_toolbar_buttons: {call: true},
+            });
+
+            window.click2Dial = {
+                custom_button_color: "orange",
+                custom_frame_color: "black",
+                dial_pad: "true",
+                did: "3001",
+                display_button: "false",
+                div_css_class_name: "btn-style-round-a",
+                draggable: "true",
+                incompatible_browser_configuration: "hide_widget",
+                placement: "bottom-right",
+                rating: "false",
+                ringback: "true",
+                server_url: "./audioconf",
+                show_branding: "false",
+                show_frame: "true",
+                text: "Ask",
+                use_default_button_css: "true",
+                protocol: "sip",    // 'sip' or 'xmpp'
+                sip: {domain: "192.168.1.251", server: "wss://desktop-545pc5b:7443/sip/proxy?url=ws://192.168.1.251:5066", register: false, caller_uri: "sip:[email protected]", authorization_user: "1002", password: "1234"},
+                xmpp: {domain: "meet.jit.si", server: "https://meet.jit.si/http-bind"}
+            }
+
+            const head  = document.getElementsByTagName('head')[0];
+            const script  = document.createElement('script');
+            script.src = click2Dial.server_url + '/scripts/click2dial.js';
+            script.async = false;
+            head.appendChild(script);
+
+            console.log("audioconf plugin is ready");
+        },
+
+        'overrides': {
+            ChatBoxView: {
+
+                toggleCall: function toggleCall(ev) {
+                    ev.stopPropagation();
+                    var room = Strophe.getNodeFromJid(this.model.attributes.jid).toLowerCase();
+                    console.debug("toggleCall", room);
+
+                    if (!voxbone.WebRTC.rtcSession.isEnded || voxbone.WebRTC.rtcSession.isEnded())
+                    {
+                        infoVoxbone.did = "audioconf_" + room;
+                        click2Dial.makeCall(true);
+
+                        this.showHelpMessages(["Calling " + infoVoxbone.did]);
+                    }
+                    else {
+                        this.showHelpMessages(["Active call in progress. Clear call and try again"]);
+                    }
+                }
+            }
+        }
+    });
+}));

audioconf/readme.md

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+# Audio Conference plugin for converse.js
+
+<img src="https://github.com/conversejs/community-plugins/blob/master/audioconf/audioconf.png" />
+
+## Overview
+This plugin adds an audio conference feature a chat or groupchat. Both SIP and XMPP protocols are supported and a server-side conference bridge is required. For SIP, a PBX like Asterisk or FreeSWITCh can be used and for XMPP, a Colibri media server like Jitsi Videobridge is needed.
+
+
+## Install
+See https://m.conversejs.org/docs/html/plugin_development.html on how to install this plugin
+
+## Configure
+To configure, edit the audioconf.js plugin file and modify the click2Dial object
+
+```
+window.click2Dial = {
+    custom_button_color: "orange",
+    custom_frame_color: "black",
+    dial_pad: "true",
+    did: "3001",
+    display_button: "false",
+    div_css_class_name: "btn-style-round-a",
+    draggable: "true",
+    incompatible_browser_configuration: "hide_widget",
+    placement: "bottom-right",
+    rating: "false",
+    ringback: "true",
+    server_url: "./audioconf",
+    show_branding: "false",
+    show_frame: "true",
+    text: "Ask",
+    use_default_button_css: "true",
+    protocol: "xmpp",    // 'sip' or 'xmpp'
+    sip: {domain: "192.168.1.251", server: "wss://desktop-545pc5b:7443/sip/proxy?url=ws://192.168.1.251:5066", register: false, caller_uri: "sip:[email protected]", authorization_user: "1002", password: "1234"},
+    xmpp: {domain: "meet.jit.si", server: "https://meet.jit.si/http-bind"}
+}
+```
+
+Default setting will use the public meet.ji.si service to provide an XMPP (colibri) type audio conference. To have a SIP based audio conference with Asterisk or FreeSWITCH, change the protocol value to "sip" and edit the sip object accordingly.
+
+## How to use
+Click on the telephone icon on the conversation toolbar to display the modal form