[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. 
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https://issues.asterisk.org/view.php?id=16896 
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Reported By:                dotnet
Assigned To:                
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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:              
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Date Submitted:             2010-02-24 09:12 CST
Last Modified:              2010-02-24 16:43 CST
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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

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 (0118520) dotnet (reporter) - 2010-02-24 16:43
 https://issues.asterisk.org/view.php?id=16896#c118520 
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I can't see how I could reproduce.

There was only 1 call at this time with 34 endpoints on the server and 1
trunk registration.

I have the SIP trace from the proxy that asterisk sent the call to.

I can upload that if it's any good to you.. Is it possible to upload it as
non public?

Thanks
Stephen 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-24 16:43 dotnet         Note Added: 0118520                          
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