[asterisk-bugs] [Asterisk 0013986]: The caller id is ignored when transferring the call

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Dec 11 15:51:25 CST 2008


The following issue has been CLOSED 
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http://bugs.digium.com/view.php?id=13986 
====================================================================== 
Reported By:                falves11
Assigned To:                putnopvut
====================================================================== 
Project:                    Asterisk
Issue ID:                   13986
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 159571 
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 reopened
Fixed in Version:           
====================================================================== 
Date Submitted:             2008-11-29 02:02 CST
Last Modified:              2008-12-11 15:51 CST
====================================================================== 
Summary:                    The caller id is ignored when transferring the call
Description: 
I need to replace the caller id before I do the transfer, so the new call
has a new caller id, bit my efforts are completely ignored. Additionally, I
try to use the function sip_addheader to add a "From:" header and that gets
ignored as well. When the call gets tranfered the "From" header has the
exact value that it had when it arrived to the Asterisk.
====================================================================== 

---------------------------------------------------------------------- 
 (0096275) blitzrage (administrator) - 2008-12-11 15:51
 http://bugs.digium.com/view.php?id=13986#c96275 
---------------------------------------------------------------------- 
I'm closing this issue as this is a feature request that does not have a
patch associated with it from the original reporter. In addition, I'm
concerned that by changing the From header being changed in an INVITE
transaction could cause other end points to basically freak out. In
addition, they may not even use it.

If you can provide some code that will provide the functionality you
desire, then I would encourage you to submit it here for review and
possible inclusion in future versions of Asterisk. If you are unable to do
the coding yourself, there are various consultants who can perform this for
you, or via the bounty system.

If you still feel this is incorrect, then please join #asterisk-bugs on
the IRC network irc.freenode.net to discuss, at which point if it is deemed
required, the issue will be reopened. Please do not reopen prior to
discussing. Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-11 15:51 blitzrage      Note Added: 0096275                          
2008-12-11 15:51 blitzrage      Status                   assigned => closed  
======================================================================




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