[asterisk-bugs] [Asterisk 0011930]: sip reload should not unregister tcp/tls peers

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Feb 5 14:47:06 CST 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11930 
====================================================================== 
Reported By:                pj
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11930
Category:                   Channels/chan_sip/Registration
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 102037 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             02-05-2008 14:35 CST
Last Modified:              02-05-2008 14:47 CST
====================================================================== 
Summary:                    sip reload should not unregister tcp/tls peers
Description: 
after reloading sip configuration, peers that using tcp or tls transport
are unregistered (even in case, when sip.conf isn't changed). 
peers are registered back when registry timeout expires. I think:
1) should not be unregistered when reloading sip.conf
2) when tcp session is closed (eg. stop/start asterisk server), tcp
clients should attempt to reregister immediatelly (not wait until
registration expires)


====================================================================== 

---------------------------------------------------------------------- 
 pj - 02-05-08 14:47  
---------------------------------------------------------------------- 
debug attached...
maybe source of the issue is, that after sip reload, asterisk server is
trying to send sip qualify message via udp, instead of tcp/tls? 

after sip reload:
OPTIONS sip:s at 192.168.40.4:5061;transport=TLS SIP/2.0
Via: SIP/2.0/UDP 192.168.38.20:5060;branch=z9hG4bK51c7964e;rport

after peer reregisters back:
OPTIONS sip:s at 192.168.40.4:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.38.20:5060;branch=z9hG4bK3671708c;rport 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
02-05-08 14:47  pj             Note Added: 0081723                          
======================================================================




More information about the asterisk-bugs mailing list