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

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Jan 13 03:03:54 CST 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14218 
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Reported By:                Nick_Lewis
Assigned To:                otherwiseguy
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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:              
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Date Submitted:             2009-01-12 10:14 CST
Last Modified:              2009-01-13 03:03 CST
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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
====================================================================== 

---------------------------------------------------------------------- 
 (0097563) Nick_Lewis (reporter) - 2009-01-13 03:03
 http://bugs.digium.com/view.php?id=14218#c97563 
---------------------------------------------------------------------- 
I thought initreqprep was called when the dialog was first created. Even if
initreqprep is called during the dialog I think p->owner->cid.cid_name is
unchanged so could be used directly each time.

However I agree that there is no benefit in testing to confirm this. I was
more interested in adding peer->fromname than removing pvt->fromname. If
this is a no-go then I will accept defeat and instead get freepbx changed
to support cid name on trunks 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-13 03:03 Nick_Lewis     Note Added: 0097563                          
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