rbrindley: branch 2.0 r4774 - in /branches/2.0/config/js: pbx2.js welcome.js

SVN commits to the Asterisk-GUI project asterisk-gui-commits at lists.digium.com
Tue May 5 14:13:46 CDT 2009


Author: rbrindley
Date: Tue May  5 14:13:43 2009
New Revision: 4774

URL: http://svn.asterisk.org/svn-view/asterisk-gui?view=rev&rev=4774
Log:

- fixed an issue where the analog filter in extensions of System Status would also show SIP/IAX users
- fixed an issue where voicemail groups wasn't properly writing to Asterisk


Modified:
    branches/2.0/config/js/pbx2.js
    branches/2.0/config/js/welcome.js

Modified: branches/2.0/config/js/pbx2.js
URL: http://svn.asterisk.org/svn-view/asterisk-gui/branches/2.0/config/js/pbx2.js?view=diff&rev=4774&r1=4773&r2=4774
==============================================================================
--- branches/2.0/config/js/pbx2.js (original)
+++ branches/2.0/config/js/pbx2.js Tue May  5 14:13:43 2009
@@ -2048,7 +2048,7 @@
 
 	var actions = new listOfSynActions('extensions.conf');
 	for (var i=0; i<lines.length; i++) {
-		actions.new_action('append', ASTGUI.contexts.VoiceMailGroups, 'exten', line[i]);
+		actions.new_action('append', ASTGUI.contexts.VoiceMailGroups, 'exten', lines[i]);
 	}
 
 	var resp = actions.callActions();

Modified: branches/2.0/config/js/welcome.js
URL: http://svn.asterisk.org/svn-view/asterisk-gui/branches/2.0/config/js/welcome.js?view=diff&rev=4774&r1=4773&r2=4774
==============================================================================
--- branches/2.0/config/js/welcome.js (original)
+++ branches/2.0/config/js/welcome.js Tue May  5 14:13:43 2009
@@ -22,7 +22,7 @@
 var REGISTRY_OUTPUT = {};
 var manager_events = {};
 var manager_timers = {};
-var extension_loads = { all: true, analog: true, features: true, iax: true, sip: true };
+var extension_loads = { all: true, analog: false, features: false, iax: false, sip: false };
 var mgr = {};
 
 var loadTrunks = function() {
@@ -123,6 +123,7 @@
 		var tmp_usertype_a = [];
 		var new_row = $('<tr></tr>');
 
+		/* check to see if the extension has iax and sip */
 		if( ud.getProperty('hassip').isAstTrue() && ud.getProperty('hasiax').isAstTrue() ) {
 			tmp_usertype_a.push( 'SIP/IAX User' );
 			new_row.addClass('iax');
@@ -133,7 +134,10 @@
 		} else if ( ud.getProperty('hassip').isAstTrue() ) {
 			tmp_usertype_a.push( '&nbsp;SIP User' );
 			new_row.addClass('sip');
-		} else if( ud.getProperty(top.sessionData.DahdiChannelString) ) {
+		}
+
+		/* check seperately if it has analog */
+		if (ud.getProperty(top.sessionData.DahdiChannelString) !== '') {
 			tmp_usertype_a.push( 'Analog User (Port ' + ud[top.sessionData.DahdiChannelString] + ')' ) ;
 			new_row.addClass('analog');
 		}
@@ -146,9 +150,9 @@
 		}
 
 		if( !ud.getProperty('context') || ! parent.sessionData.pbxinfo.callingPlans[ud.getProperty('context')] ){
-			var tmp_userstring = '<u>' + user + '</u> <font color=red>*No DialPlan assigned</font>' ;
+			var tmp_userstring = user + '<font color=red>*No DialPlan assigned</font>' ;
 		}else{
-			var tmp_userstring = '<u>' + user + '</u>' ;
+			var tmp_userstring = user;
 		}
 
 		$("<td></td>").html(ASTGUI.getUser_DeviceStatus_Image(user)).attr("id","exten_status_"+user).appendTo(new_row);




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