[asterisk-bugs] [Asterisk 0016896]: Asterisk Crashes after it thinks it gets corrupt SIP message

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16896 
====================================================================== 
Reported By:                dotnet
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16896
Category:                   Channels/chan_sip/General
Reproducibility:            unable to reproduce
Severity:                   crash
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.29.1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-02-24 09:12 CST
Last Modified:              2010-02-24 16:16 CST
====================================================================== 
Summary:                    Asterisk Crashes after it thinks it gets corrupt SIP
message
Description: 
I had call from Customer - There phone resgistrations had failed..

I checked the CLI and as soon as I typed SIP SHOW [TAB] the CLI hungup

I restarted asterisk /etc/init.d/asterisk restart and it came back fine. 

I checked the asterisk logs and the last entry was

[Feb 24 13:58:22] WARNING[20879] chan_sip.c: Bad request protocol
H^?	x??	10044 at x.x.x.x:5060 SIP/2.0 (ip x'd)

I have been running 1.4.21 since last September with NEVER a crash. I
upgraded (sic) to Asterisk 1.4.29.1 built by root @ carl on a i686 last
night because I had a requirement to get dahdi installed on the system.

The call that crashed * went through our SIP proxy (Kamailio) - the SIP
messages look fine..

What more information should I give you?  Should I regress to 1.4.21 to
provide a stable platform for my customers.

Regards.
Stephen

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

---------------------------------------------------------------------- 
 (0118518) junky (manager) - 2010-02-24 16:16
 https://issues.asterisk.org/view.php?id=16896#c118518 
---------------------------------------------------------------------- 
Could you reproduce this easily?

You should include the SIP debug.
When you say "the SIP messages look fine.." do you have it?

Many simultaneous calls? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-02-24 16:16 junky          Note Added: 0118518                          
======================================================================