[asterisk-bugs] [Asterisk 0019182]: [regression] Asterisk drops sip messages and/or response codes if SIP/TLS is used

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=19182 
====================================================================== 
Reported By:                st
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   19182
Category:                   Channels/chan_sip/TCP-TLS
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.8.3.3 
JIRA:                       SWP-3404 
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-04-26 10:11 CDT
Last Modified:              2011-05-13 02:26 CDT
====================================================================== 
Summary:                    [regression] Asterisk drops sip messages and/or
response codes if SIP/TLS is used
Description: 
When a Snom 360 (Firmware 7.3.7) tries to register, there is a new tcp
connection. Sometimes Asterisk CLI shows Register and 401 Response, but the
phone does not get a response. Sometimes CLI shows nothing, but the phone
logs many attempts to register. Its strange and not easy to reproduce, but
here sipp is the solution.


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

---------------------------------------------------------------------- 
 (0134878) kasomaro (reporter) - 2011-05-13 02:26
 https://issues.asterisk.org/view.php?id=19182#c134878 
---------------------------------------------------------------------- 
The new 1.8.4 TLS/SRTP looks like to be broken too. 1.8.3.2 is working
perfect (same config). How can I get more log from the TLS process, which
results in: "SIP/2.0 401 Unauthorized"? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-05-13 02:26 kasomaro       Note Added: 0134878                          
======================================================================