[asterisk-bugs] [Asterisk 0019190]: When calling myself with a SIP TRUNK with ITSP provider, incoming call is considered diverted and channel converted to Local/

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=19190 
====================================================================== 
Reported By:                albersag
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   19190
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   trivial
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.17.3 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-04-27 10:35 CDT
Last Modified:              2011-04-27 11:15 CDT
====================================================================== 
Summary:                    When calling myself with a SIP TRUNK with ITSP
provider, incoming call is considered diverted and channel converted to Local/
Description: 
Asterisk connected to SIP Provider, seems to bridge SIP Channels or Console
Channels as a call forwarding when they are incoming INVITES related to
outbound call made with same SIP provider account. ( I call myself).

When asterisk make this threatment, i could not use SIP functions s as
SIP_HEADER, because it is not a SIP CHANNEL as it has been considered a
call diverted.

I attach sip trace
====================================================================== 

---------------------------------------------------------------------- 
 (0134192) albersag (reporter) - 2011-04-27 11:15
 https://issues.asterisk.org/view.php?id=19190#c134192 
---------------------------------------------------------------------- 
Please take a look to second trace. Sorry for last RTF format on file.

This call has been done from SIP local extension and i get the same

  -- Now forwarding SIP/100-000004bd to 'Local/s at from-voztele' (thanks to
SIP/NEXTC-000004be) 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-04-27 11:15 albersag       Note Added: 0134192                          
======================================================================