[asterisk-bugs] [Asterisk 0013424]: r140417 broke sip nat interoperability

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Sep 5 10:51:08 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13424 
====================================================================== 
Reported By:                mdu113
Assigned To:                putnopvut
====================================================================== 
Project:                    Asterisk
Issue ID:                   13424
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 141028 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-09-04 15:35 CDT
Last Modified:              2008-09-05 10:51 CDT
====================================================================== 
Summary:                    r140417 broke sip nat interoperability
Description: 
The scenario is very simple. I have 2 sip endpoints: xyz011101 and
xyz010001, both on a private LAN (192.168.0.0/24) behind the same nat
device (cisco router) and asterisk on a public ip. This setup has been
working for years, but after changes made in r140417, asterisk seems to
ignore the source port the device was registered from and always sends
requests to registration_ip:5060, which of course doesn't work.
====================================================================== 

---------------------------------------------------------------------- 
 (0092109) svnbot (reporter) - 2008-09-05 10:51
 http://bugs.digium.com/view.php?id=13424#c92109 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 141217

U   branches/1.4/channels/chan_sip.c

------------------------------------------------------------------------
r141217 | mmichelson | 2008-09-05 10:51:07 -0500 (Fri, 05 Sep 2008) | 14
lines

Commit 140417 had a logic flaw in it which
caused port 5060 to always be used when dialing
a peer if no explicit port was specified. This
broke the behavior of implicitly using the port
from which the peer registered if no port is
specified. This commit fixes the logic flaw.

(closes issue http://bugs.digium.com/view.php?id=13424)
Reported by: mdu113
Patches:
      13424.patch uploaded by putnopvut (license 60)
Tested by: mdu113


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

http://svn.digium.com/view/asterisk?view=rev&revision=141217 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-09-05 10:51 svnbot         Checkin                                      
2008-09-05 10:51 svnbot         Note Added: 0092109                          
======================================================================




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