[asterisk-bugs] [Asterisk 0011322]: Respect the original "From" header in "Dial" application

[noreply at bugs.digium.com]

The following issue has been CLOSED 
====================================================================== 
http://bugs.digium.com/view.php?id=11322 
====================================================================== 
Reported By:                ibc
Assigned To:                oej
====================================================================== 
Project:                    Asterisk
Issue ID:                   11322
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.14  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 open
Fixed in Version:           
====================================================================== 
Date Submitted:             11-20-2007 05:57 CST
Last Modified:              01-22-2008 09:59 CST
====================================================================== 
Summary:                    Respect the original "From" header in "Dial"
application
Description: 
If Asterisk receives an incoming SIP call from an **external** SIP account,
and during the dialplan it must call to an internal SIP user, the INVITE
from Asterisk has a modified "From" header (the fromdomain is changed by
Asterisk). Why?

This is not needed and it's very anti-RFC ("From" shouldn't be changed).

And in fact it can cause problems, as this example:

- Our Asterisk domain is "asterisk_domain.org".

- We have two internal users:
  - "Bob" <200>
  - "Pepe" <201>

- We receive an external SIP incoming call with this From header:
  From: "EXTERNAL <sip:201 at external_domain.com>"

- During the dialplan Asterisk finally does:
  Dial(SIP/200)

- This new INVITE generated by Asterisk will contain this header:
  From: "EXTERNAL" <sip:201 at asterisk_domain.org>   <--- DOMAIN CHANGED
!!!!


So our internal user "Bob" <200> will receive a call and will think that
it's a call from our internal user "Pepe" <201>. If, for example, the call
is not answered and later Bob open him "missed calls" he will see a call
from 201, and if he calls he will call in fact to "Pepe <201>".

This is IMHO a wrong behaviour. Asterisk shoudn't change the "From"
header, there is no reason for that. Why a internal user can't know from
which domain he recives a call vía Asterisk?

For the established dialog the "From" domain is not important at all, the
only the called needs to know if the Call-ID, From/To tags and Contact
header to send in-dialog requests, no more.

I attach a debug example that shows how the "From" header is changed in
the INVITE generated by Asterisk.
====================================================================== 

---------------------------------------------------------------------- 
 oej - 01-22-08 09:59  
---------------------------------------------------------------------- 
Closing this for now. Continuing work on the branch and trying to find a
long term fix for this issue. Thanks for your patience and support! 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-22-08 09:59  oej            Status                   assigned => closed  
01-22-08 09:59  oej            Note Added: 0081024                          
======================================================================