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

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Dec 9 17:47:52 CST 2008


The following issue has been UPDATED. 
====================================================================== 
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:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             2008-11-29 02:02 CST
Last Modified:              2008-12-09 17:47 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.
====================================================================== 

---------------------------------------------------------------------- 
 (0096057) putnopvut (administrator) - 2008-12-09 17:47
 http://bugs.digium.com/view.php?id=13986#c96057 
---------------------------------------------------------------------- 
I re-read note http://bugs.digium.com/view.php?id=13986#c95825. This confirms
that you are trying to transfer the
call away from Asterisk. In this case, it is absolutely impossible for the
caller id to be propagated to the new call. I know of no way to do that
based on any of the SIP RFCs or drafts that I have read. This is most
definitely a feature request and not a bug.

If you need the caller id to be changed for outbound calls that do not go
through Asterisk, the only way that I can think to do so would be to add a
SIP proxy to the path to do the job for you.

Closing. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-09 17:47 putnopvut      Note Added: 0096057                          
2008-12-09 17:47 putnopvut      Status                   assigned => resolved
2008-12-09 17:47 putnopvut      Resolution               reopened => fixed   
2008-12-09 17:47 putnopvut      Assigned To               => putnopvut       
2008-12-09 17:47 putnopvut      Status                   resolved => closed  
======================================================================




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