[asterisk-bugs] [Asterisk 0012215]: [patch] Asterisk returns 482 Loop Detected upon receiving re-invite

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12215 
====================================================================== 
Reported By:                jpyle
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   12215
Category:                   Channels/chan_sip/General
Reproducibility:            random
Severity:                   minor
Priority:                   normal
Status:                     confirmed
Asterisk Version:           1.4.18 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-03-14 11:45 CDT
Last Modified:              2008-11-14 09:06 CST
====================================================================== 
Summary:                    [patch] Asterisk returns 482 Loop Detected upon
receiving re-invite
Description: 
Asterisk sends a 482 Loop Detected upon receiving a presumably valid
re-INVITE.  Pedantic is enabled globally in sip.conf.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
duplicate of        0007403 [patch] allow SIP Spiral to work instea...
====================================================================== 

---------------------------------------------------------------------- 
 (0094899) remiq (reporter) - 2008-11-14 09:06
 http://bugs.digium.com/view.php?id=12215#c94899 
---------------------------------------------------------------------- 
You could be right it could expose a bug, but what is wrong with that trace
is that the far end hung up the call and asterisk still shows the channel
up.  Also, the fact that you never see the 200 OK when the call gets
answered in the logs.  Asterisk clearly answered the call and re-invited
properly I verified that, but the logs don't show that at all.  All I see
is this:

<------------>    -- SIP/fsa-fsdev-0a1d7da8 answered Zap/1-1

No SIP debug message showing 200 OK or even the re-INVITE.  And like I
said before even when the far end hangs up nothing happens, the calling
party has to hang up in order for asterisk to hang the call up.  So
something is getting messed up.  I think more code has to be written when
the channel is not marked UP and when asterisk is processing a re-INVITE
because the call is not getting setup properly with the patch you are
proposing. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-11-14 09:06 remiq          Note Added: 0094899                          
======================================================================