pari: branch 2.0 r4389 - in /branches/2.0/config: js/tooltip.js users.html

[SVN commits to the Asterisk-GUI project]

Author: pari
Date: Mon Dec 22 16:50:58 2008
New Revision: 4389

URL: http://svn.digium.com/view/asterisk-gui?view=rev&rev=4389
Log:

 fixing tooltip for line keys



Modified:
    branches/2.0/config/js/tooltip.js
    branches/2.0/config/users.html

Modified: branches/2.0/config/js/tooltip.js
URL: http://svn.digium.com/view/asterisk-gui/branches/2.0/config/js/tooltip.js?view=diff&rev=4389&r1=4388&r2=4389
==============================================================================
--- branches/2.0/config/js/tooltip.js (original)
+++ branches/2.0/config/js/tooltip.js Mon Dec 22 16:50:58 2008
@@ -66,6 +66,8 @@
 	tooltips['users'] .en[30] = "<B>Flash</B> sets the amount of time, in milliseconds, that must have passed since the last hook-flash event received by asterisk before it will recognize a second event.  If a second event occurs in less time than defined for Flash, then asterisk will ignore the event.  The default value of Flash is 750 ms, and it can be configured in 1ms increments." ;
 
 	tooltips['users'] .en[31] = "<B>RXFlash</B> sets the amount of time, in milliseconds, that the hook-flash must remain depressed in order for asterisk to consider such an event a valid flash event.  The default value of RXFlash is 1250ms and it can be configured in 1ms increments." ;
+
+	tooltips['users'] .en[94] = "Line Keys is the number of lines that are tied to this SIP registration.";
 
 	tooltips['users'] .en[95] = "<B>Codecs</B> A codec is a compression or decompression algorithm run against voice as it is moved between analog (speaking) and digital (VoIP). <B>u-law</B> A PSTN standard codec, used in North America, that provides very good voice quality and consumes 64kbit/s for each direction (receiving and transmitting) of a VoIP call.  u-law should be supported by all VoIP phones. <B>a-law</B> A PSTN standard codec, used outside of North America, that provides very good voice quality and consumes 64kbit/s for each direction (receiving and transmitting) of a VoIP call.  a-law should be supported by all VoIP phones. <B>GSM</B> A wireless standard codec, used worldwide, that provides okay voice quality and consumes 13.3kbit/s for each direction (receiving and transmitting) of a VoIP call.  GSM is supported by many VoIP phones. <B>G.726</B> A PSTN codec, used worldwide, that provides good voice quality and consumes 32kbit/s for each direction (receiving and 
 transmitting) of a VoIP call. G.726 is supported by some VoIP phones. <B>G.722</B> A high-fidelity codec for VoIP calls that provides excellent voice quality and consumes 64kbit/s for each direction (receiving and transmitting) of a VoIP call. At present, G.722 is only supported by a limited number of VoIP phones.";
 

Modified: branches/2.0/config/users.html
URL: http://svn.digium.com/view/asterisk-gui/branches/2.0/config/users.html?view=diff&rev=4389&r1=4388&r2=4389
==============================================================================
--- branches/2.0/config/users.html (original)
+++ branches/2.0/config/users.html Mon Dec 22 16:50:58 2008
@@ -184,7 +184,7 @@
 								<option value='5'>5</option>
 								<option value='6'>6</option>
 							</select>
-							<img src="images/tooltip_info.gif" tip="en,users,2" class='tooltipinfo'>
+							<img src="images/tooltip_info.gif" tip="en,users,94" class='tooltipinfo'>
 							
 							SIP/IAX Password: <input id="edit_secret" size=6>
 							<img src="images/tooltip_info.gif" tip="en,users,2" class='tooltipinfo'>