[asterisk-bugs] [Asterisk 0014575]: BYE to 408 Request Timeout

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=14575 
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Reported By:                chris-mac
Assigned To:                mmichelson
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Project:                    Asterisk
Issue ID:                   14575
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 178608 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2009-03-01 06:59 CST
Last Modified:              2009-07-24 13:31 CDT
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Summary:                    BYE to 408 Request Timeout
Description: 
Noticed * is generating BYE request to "408 Request Timeout" response.
I don't think this is correct, as 4xx,5xx and 6xx responses to INVITE
should
be only acknowledged with ACK (RFC3261 - 13.2.2.3).

Proxy                                 Asterisk
|<-----------------------(sdp) INVITE F8<|
|>F9 100 Giving a try ------------------>|
|>F13 408 Request Timeout -------------->|
|<------------------------------ ACK F14<|
|<------------------------------ BYE F15<|  <= THIS SHOULD NOT BE HERE
|>F18 404 Not here --------------------->|

Please see 408-flow.html attached for detailed call flow.

How to reproduce:
1. You will need a SIP proxy (eg. OpenSER)
2. Edit sip.conf attached and modify <ASTERISK_IP>, <PROXY_IP>,
<DEAD_HOST>
3. Start * with sip.conf and extensions.conf attached.
4. Dial into * with any SIP phone


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---------------------------------------------------------------------- 
 (0108193) svnbot (reporter) - 2009-07-24 13:31
 https://issues.asterisk.org/view.php?id=14575#c108193 
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Repository: asterisk
Revision: 208589

_U  branches/1.6.0/
U   branches/1.6.0/channels/chan_sip.c

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r208589 | mmichelson | 2009-07-24 13:31:41 -0500 (Fri, 24 Jul 2009) | 23
lines

Merged revisions 208588 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) | 16
lines
  
  Merged revisions 208587 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) |
10 lines
    
    Only send a BYE when hanging up a channel that is up.
    
    For cases where Asterisk sends an INVITE and receives a non 2XX final
    response, Asterisk would follow the INVITE transaction by immediately
    sending a BYE, which was unnecessary.
    
    (closes issue https://issues.asterisk.org/view.php?id=14575)
    Reported by: chris-mac
  ........
................

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http://svn.digium.com/view/asterisk?view=rev&revision=208589 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-24 13:31 svnbot         Checkin                                      
2009-07-24 13:31 svnbot         Note Added: 0108193                          
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