[asterisk-bugs] [Asterisk 0012169]: sip debug can't be stopped

noreply at bugs.digium.com noreply at bugs.digium.com
Sat Mar 15 04:05:59 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12169 
====================================================================== 
Reported By:                pj
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   12169
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 105677 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             03-07-2008 13:56 CST
Last Modified:              03-15-2008 04:05 CDT
====================================================================== 
Summary:                    sip debug can't be stopped
Description: 
when turn on sip debug globally or on per peer basis, it can't be stopped,
it seems, that affected are only tls peers

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

---------------------------------------------------------------------- 
 pj - 03-15-08 04:05  
---------------------------------------------------------------------- 
when sp debug is turned on, "sip show settings" shows:
  SIP Debug: 2
when do "sip set debug off" it shows:
  SIP Debug: 0
but cli output still continues to show debug messages (below) from tls
peers...
I haven't any "sipdebug" setting in sip.conf.
running: SVN-trunk-r105677 + bug11972-8.diff.txt


[Mar 15 10:01:33]
<--- SIP read from TLS://85.216.201.137:37598 --->
REGISTER sip:ipbx.i.cz:5061 SIP/2.0
Via: SIP/2.0/TLS 85.216.201.137:0;branch=z9hG4bK78f33702;rport
Max-Forwards: 70
From: <sip:726 at ipbx.i.cz>;tag=as3ded5622
To: <sip:726 at ipbx.i.cz>
Call-ID: 1fc017fb1c335ab92901893903a7d9e4 at 10.0.0.2
CSeq: 970 REGISTER
User-Agent: Asterisk PBX SVN-trunk-r103314
Authorization: Digest username="726", realm="ipbx.i.cz", algorithm=MD5,
uri="sip:ipbx.i.cz:5061", nonce="1bd3742a",
response="46aca21ca22004bee7f000ca57db9ace", opaque=""
Expires: 120
Contact: <sip:726 at 85.216.201.137:5061;transport=TLS>
Event: registration
Content-Length: 0


<------------->
[Mar 15 10:01:33] --- (13 headers 0 lines) ---
[Mar 15 10:01:33]
<--- SIP read from TLS://85.216.201.137:37598 --->
REGISTER sip:ipbx.i.cz:5061 SIP/2.0
Via: SIP/2.0/TLS 85.216.201.137:0;branch=z9hG4bK2c29178f;rport
Max-Forwards: 70
From: <sip:726 at ipbx.i.cz>;tag=as01fca516
To: <sip:726 at ipbx.i.cz>
Call-ID: 1fc017fb1c335ab92901893903a7d9e4 at 10.0.0.2
CSeq: 971 REGISTER
User-Agent: Asterisk PBX SVN-trunk-r103314
Authorization: Digest username="726", realm="ipbx.i.cz", algorithm=MD5,
uri="sip:ipbx.i.cz:5061", nonce="1c167fb6",
response="eab27a47b492cb1556e2a7a91a3f7f2d", opaque=""
Expires: 120
Contact: <sip:726 at 85.216.201.137:5061;transport=TLS>
Event: registration
Content-Length: 0


<------------->
[Mar 15 10:01:33] --- (13 headers 0 lines) --- 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
03-15-08 04:05  pj             Note Added: 0084006                          
======================================================================




More information about the asterisk-bugs mailing list