[asterisk-bugs] [Asterisk 0014218]: [patch] Not possible to disguise display name on calls to trunks even though user can be disguised

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14218 
====================================================================== 
Reported By:                Nick_Lewis
Assigned To:                otherwiseguy
====================================================================== 
Project:                    Asterisk
Issue ID:                   14218
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     feedback
Target Version:             1.6.2
Asterisk Version:           1.6.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2009-01-12 10:14 CST
Last Modified:              2009-01-13 10:17 CST
====================================================================== 
Summary:                    [patch] Not possible to disguise display name on
calls to trunks even though user can be disguised
Description: 
When calling from an extension on the asterisk PBX to a trunk the sip
display-name that is received in the invite from the extension is used as
the display-name in the invite sent to the ITSP.

For example:
 From: "John Smith" <sip:101 at mypbx>
is converted to:
 From: "John Smith" <sip:voipaccount at myitsp>
but it would be useful if it could be converted instead to:
 From: "My Corporation" <sip:voipaccount at myitsp>
so that outgoing calls do not reveal the caller name
====================================================================== 

---------------------------------------------------------------------- 
 (0097574) otherwiseguy (administrator) - 2009-01-13 10:17
 http://bugs.digium.com/view.php?id=14218#c97574 
---------------------------------------------------------------------- 
That would probably be the best way to go.  We try to repeat to ourselves
"Asterisk is multi-protocol PBX" when looking at changes for inclusion.  If
there is a way to handle something for all channel types, we tend to prefer
that to changes for individual channels.

BTW, if you are the Nick Lewis that contributes to freepbx and drupal (or
maybe those are different Nick Lewis'?), "Hi!" and "Thanks!" 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-13 10:17 otherwiseguy   Note Added: 0097574                          
======================================================================