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

noreply at bugs.digium.com noreply at bugs.digium.com
Sun Feb 17 12:39:29 CST 2008


A NOTE has been added to this issue. 
====================================================================== 
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:              02-17-2008 12:39 CST
====================================================================== 
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.
====================================================================== 

---------------------------------------------------------------------- 
 jmls - 02-17-08 12:39  
---------------------------------------------------------------------- 
Reporter has stated that he has sent his disclaimer in. Can someone at
Digium please check the licence status for him.

Thanks. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
02-17-08 12:39  jmls           Note Added: 0082385                          
======================================================================




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