[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:              2009-01-12 11:13 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
====================================================================== 

---------------------------------------------------------------------- 
 (0097504) otherwiseguy (administrator) - 2009-01-12 11:13
 http://bugs.digium.com/view.php?id=14218#c97504 
---------------------------------------------------------------------- 
It looks like fromname is first set from n, which is originally set to
p->owner->cid.cid_name, which could be set from the CALLERID(name).

Nick_Lewis, could you try adding a Set(CALLERID(name)=My Coproration to
your outbound dialing and see if that works for you?  I can test here, but
would have to set up a second Asterisk box.  But the code looks like
fromuser will only be set via the channel cid, which will initially be set
by the callerid= line in sip.conf, after that, it is up to you to use the
CALLERID() dialplan function. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-12 11:13 otherwiseguy   Note Added: 0097504                          
======================================================================