[asterisk-bugs] [Asterisk 0012799]: fresh reboot of asterisk server, TCP sip peers get invite via UDP

noreply at bugs.digium.com noreply at bugs.digium.com
Fri Jun 27 11:21:04 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12799 
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Reported By:                pabelanger
Assigned To:                bbryant
====================================================================== 
Project:                    Asterisk
Issue ID:                   12799
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0-beta9 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             06-05-2008 11:41 CDT
Last Modified:              06-27-2008 11:21 CDT
====================================================================== 
Summary:                    fresh reboot of asterisk server, TCP sip peers get
invite via UDP
Description: 
Weird issue, 

If we reboot our asterisk server (# sudo reboot), then let everything boot
back up.  SIP peers configured for TCP seem to default to UDP clients.  But
if we issue 'reload' from the asterisk cli or init.d script, the SIP peers
will be setup back to TCP.
====================================================================== 

---------------------------------------------------------------------- 
 svnbot - 06-27-08 11:21  
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 125891

U   trunk/channels/chan_sip.c
U   trunk/configs/sip.conf.sample

------------------------------------------------------------------------
r125891 | bbryant | 2008-06-27 11:20:56 -0500 (Fri, 27 Jun 2008) | 6 lines

Change the way that the transport option works for sip users. transport
will now take multiple arguments, the first one listed will be the one used

for new dialogs, and the rest listed will be acceptable ways for that peer
to contact us. This fixes a minor bug where, because SIP TCP/UDP run on 
the same port, could cause a TCP peer to be saved in the ast_db. There
will also be warnings when a transport is changed for an unexpected reason.

(issue http://bugs.digium.com/view.php?id=12799)

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

http://svn.digium.com/view/asterisk?view=rev&revision=125891 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
06-27-08 11:21  svnbot         Note Added: 0089362                          
======================================================================




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