[asterisk-bugs] [Asterisk 0009838]: Bye authorization working only one way.

noreply at bugs.digium.com noreply at bugs.digium.com
Mon Sep 10 15:42:13 CDT 2007


The following issue has been REOPENED. 
====================================================================== 
http://bugs.digium.com/view.php?id=9838 
====================================================================== 
Reported By:                absa
Assigned To:                oej
====================================================================== 
Project:                    Asterisk
Issue ID:                   9838
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:            1.2.18  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        No 
Request Review:              
====================================================================== 
Date Submitted:             05-30-2007 13:47 CDT
Last Modified:              09-10-2007 15:42 CDT
====================================================================== 
Summary:                    Bye authorization working only one way.
Description: 
When hanging up calls, one side is always left hanging with no sound signal
and call doesn't end for the receiver.
There are two scenarios and in one bug always occurs, and in opposite
scenario BYE authorizes as it should.

(->) indicates call route.

First Scenarion, incoming call (works ok):
Caller -> [SIP_provider_with_auth] -> asterisk -> SIPphone (makes the
hangup).

Second scenarion, outgoing call (with bug):
Receiver <- [SIP_provider_with_auth] <- asterisk <- SIPphone (makes the
hangup).

In both scenarios SIP provider gets BYE request, sends 401 Unauthorized,
in first scenarion receives authentication, and in second scenario doesn't
receive authentication on BYE request and leaves the call hanging at SIP
registrars side. Also in both scenarios hanging up is made with the SIP
phone, that is peered  with asterisk.

I think this bug is the same or related to:
http://bugs.digium.com/view.php?id=9681

I have tried it with 1.2.18 clean compiled from source with no patches,
and with  SVN-branch-1.2-r66537M, in both cases bug exists.
====================================================================== 

---------------------------------------------------------------------- 
 absa - 09-10-07 15:42  
---------------------------------------------------------------------- 
Reproduced the bug in 1.4. 

I have provided a patch for 1.2, and will provide a patch for 1.4 ASAP. I
have also sent everything that is needed for patch to be merged (including
a fax with my legal consent), so i can't understand why is it taking so
long? 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
09-10-07 15:42  absa           Status                   closed => feedback  
09-10-07 15:42  absa           Resolution               won't fix => reopened
09-10-07 15:42  absa           Note Added: 0070288                          
======================================================================




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