pari: trunk r1266 - in /trunk/config: scripts/tooltip.js users.html

[SVN commits to the Asterisk-GUI project]

Author: pari
Date: Fri Jul 27 11:04:05 2007
New Revision: 1266

URL: http://svn.digium.com/view/asterisk-gui?view=rev&rev=1266
Log:
Fix: some tool tips in the users tab are not present.

Modified:
    trunk/config/scripts/tooltip.js
    trunk/config/users.html

Modified: trunk/config/scripts/tooltip.js
URL: http://svn.digium.com/view/asterisk-gui/trunk/config/scripts/tooltip.js?view=diff&rev=1266&r1=1265&r2=1266
==============================================================================
--- trunk/config/scripts/tooltip.js (original)
+++ trunk/config/scripts/tooltip.js Fri Jul 27 11:04:05 2007
@@ -38,7 +38,9 @@
 	tooltips['users'] .en[18] = "<B>Agent Login Extension:</B> Extension to be dialed for the Agents to Login to the Specific Queue. <br> This is an extension that all the Agents can Call to Login to their specified Queues. ";
 	tooltips['users'] .en[19] = "<B>Agent Callback Login Extension:</B> Extension to be dialed for the Agents to Login to the Queues they are apart of.<br> Same as Agent Login Extension, except you do not have to remain on the line. ";
 	tooltips['users'] .en[20] = "<B>Agent Logout</B><BR><LI> To logout of <b>Agent Login</b> Hangup your phone. <LI>To Logout of <b>Agent Callback Login</b> Dial the same extension used to login, specify your extension and password when prompted, and hit # when asked for your callback extension. This will successfully log you out of all queues you are apart of.";
-	
+	tooltips['users'] .en[21] = "<B>Can Reinvite:</B> This option can be used to tell the Asterisk server whethere or not to issue a reinvite to the client. ";
+	tooltips['users'] .en[22] = "<B>NAT:</B> Try this setting when Asterisk is on a public IP, communicating with devices hidden behind a NAT device (broadband router). If you have one-way audio problems, you usually have problems with your NAT configuration or your firewall's support of SIP+RTP ports.";
+	tooltips['users'] .en[23] = "<B>DTMFMode:</B> Set default dtmfmode for sending DTMF. Default: rfc2833 <BR><B>Other options:</B><BR>info : SIP INFO messages<BR>inband : Inband audio (requires 64 kbit codec -alaw, ulaw)<BR>auto : Use rfc2833 if offered, inband otherwise";
 	tooltips['users'] .en[99] = "<B>Phone Serial:</B> Enter the serial number of a Polycom phone to enable phone provisioning." ;
 
 // Tooltips for Conferencing (meetme)

Modified: trunk/config/users.html
URL: http://svn.digium.com/view/asterisk-gui/trunk/config/users.html?view=diff&rev=1266&r1=1265&r2=1266
==============================================================================
--- trunk/config/users.html (original)
+++ trunk/config/users.html Fri Jul 27 11:04:05 2007
@@ -484,15 +484,15 @@
 				</tr>
 				<tr>
 					<td align=right><input type='checkbox' id='canreinvite'></td>
-					<td class="field_text">Can Reinvite</td>
+					<td class="field_text" tip="en,users,21">Can Reinvite</td>
 
 					<td align=right><input type='checkbox' id='nat'></td>
-					<td class="field_text">NAT</td>
+					<td class="field_text" tip="en,users,22">NAT</td>
 				</tr>
 
 				<tr>
 					<td align=right colspan=3><input id="dtmfmode" size=10 class="input8" dfalt="rfc2833"></td>
-					<td class="field_text">DTMFMode</td>
+					<td class="field_text" tip="en,users,23">DTMFMode</td>
 				</tr>
 				</table>
 				</fieldset>