[asterisk-bugs] [Asterisk 0018504]: SIP autocreated peers crash asterisk on call

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Dec 20 21:40:33 UTC 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18504 
====================================================================== 
Reported By:                kkm
Assigned To:                mmichelson
====================================================================== 
Project:                    Asterisk
Issue ID:                   18504
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     closed
Asterisk Version:           1.8.1.1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             2010-12-20 15:04 CST
Last Modified:              2010-12-20 15:40 CST
====================================================================== 
Summary:                    SIP autocreated peers crash asterisk on call
Description: 
So I am the only person in the world using autopeers, out in the cold
alone. Scary. Proud but scared.

Any call to any SIP autopeer crashes, by way of either Dial() or Queue().

Note that a call on other direction, to a static peer, won't crash
asterisk.
====================================================================== 

---------------------------------------------------------------------- 
 (0129812) svnbot (reporter) - 2010-12-20 15:40
 https://issues.asterisk.org/view.php?id=18504#c129812 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 299249

_U  trunk/
U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r299249 | mmichelson | 2010-12-20 15:40:33 -0600 (Mon, 20 Dec 2010) | 25
lines

Merged revisions 299248 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r299248 | mmichelson | 2010-12-20 15:38:30 -0600 (Mon, 20 Dec 2010) | 20
lines
  
  Fix a couple of CCSS issues.
  
  * Make sure to allocate a cc_params structure
    when creating autopeers.
  
  * Use sip_uri_cmp when retrieving SIP CC agents
    and monitors in case parameters appear in the
    URI.
  
  (closes issue https://issues.asterisk.org/view.php?id=18504)
  Reported by: kkm
  
  (closes issue https://issues.asterisk.org/view.php?id=18338)
  Reported by: GeorgeKonopacki
  Patches: 
        18338.diff uploaded by mmichelson (license 60)
  Tested by: GeorgeKonopacki
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=299249 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-12-20 15:40 svnbot         Checkin                                      
2010-12-20 15:40 svnbot         Note Added: 0129812                          
======================================================================




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