[asterisk-bugs] [Asterisk 0016672]: Asterisk manage forked calls as reinvite

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16672 
====================================================================== 
Reported By:                tucceri
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16672
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.29 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-01-22 09:56 CST
Last Modified:              2010-01-22 10:59 CST
====================================================================== 
Summary:                    Asterisk manage forked calls as reinvite
Description: 
Forked calls by remote server are managed as reinvites -> 2nd is rejected.

 UAC -> Remote server     +--> to lagacy phone calls +-> Asterisk         
     
       (without advanced   |                          |
       services            |                          |
       or gateway          |                          |
       to PSTN features)   |                          |
                           +----> voicemail to catch unanswered state
====================================================================== 

---------------------------------------------------------------------- 
 (0117074) lmadsen (administrator) - 2010-01-22 10:59
 https://issues.asterisk.org/view.php?id=16672#c117074 
---------------------------------------------------------------------- 
Please provide enough information that describes the call flow better along
with configuration information in order to reproduce this issue. If you're
describing an issue with how SIP is being handled, you must provide the sip
trace and sip history along with console output showing the issue as per
the bug guidelines. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-01-22 10:59 lmadsen        Note Added: 0117074                          
2010-01-22 10:59 lmadsen        Status                   new => feedback     
======================================================================