[test-results] [Bamboo] Asterisk - Trunk > Ubuntu Lucid (10.04) > #1821 has FAILED (2 tests failed, 1 failures were new). Change made by Terry Wilson and Russell Bryant.

[Bamboo]

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Asterisk - Trunk > Ubuntu Lucid (10.04) > #1821 failed.
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Code has been updated by Terry Wilson, Russell Bryant.
2/198 tests failed, 1 failure was new.

http://bamboo.asterisk.org/browse/ASTTRUNK-LUCID-1821/


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Failing Jobs
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  - amd64 (Default Stage): 2 of 198 tests failed.


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Code Changes
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Terry Wilson (351143):

>Ensure ACK retransmit & hangup on non-200 response to INVITE
>
>When handling a non-2xx final response on an INVITE transaction, we have to
>keep the transaction around after we send an ACK in case we receive a
>retransmission of the response so we can re-transmit the ACK, but also tear
>down the ast_channel as soon as we transmit the ACK. Before this patch, we
>could fail at both of these things. Calling sip_alreadygone/needdestroy
>prevented us from keeping the transaction up and retransmitting the ACK, and
>queueing CONGESTION was not sufficient to cause the channel to be torn down
>when originating calls via the CLI, for example.
>
>This patch queues a hangup with CONGESTION instead of just queueing CONGESTION
>for these responses and removes the sip_alreadygone and sip_needdestroy calls
>from handle_response_invite on non-2xx responses. It relies on the hangup
>calling sip_scheddestroy.
>
>For more information, see section 17.1.1.1 of RFC 3261.
>
>(closes issue ASTERISK-17717)
>Review: https://reviewboard.asterisk.org/r/1672/
>........
>
>Merged revisions 351130 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 351131 from http://svn.asterisk.org/svn/asterisk/branches/10
>

Russell Bryant (351184):

>Merged revisions 351183 via svnmerge from 
>https://origsvn.digium.com/svn/asterisk/branches/10
>
>................
>  r351183 | russell | 2012-01-16 20:43:19 -0500 (Mon, 16 Jan 2012) | 29 lines
>  
>  Merged revisions 351182 via svnmerge from 
>  https://origsvn.digium.com/svn/asterisk/branches/1.8
>  
>  ........
>    r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012) | 22 lines
>    
>    Add some missing locking in chan_sip.
>    
>    This patch adds some missing locking to the function 
>    send_provisional_keepalive_full().  This function is called from the scheduler,
>    which is processed in the SIP monitor thread.  The associated channel (or pbx)
>    thread will also be using the same sip_pvt and ast_channel so locking must be
>    used.  The sip_pvt_lock_full() function is used to ensure proper locking order
>    in a safe manner.
>    
>    In passing, document a suspected reference counting error in this function.
>    The "fix" is left commented out because when the "fix" is present, crashes
>    occur.  My theory is that fixing it is exposing a reference counting error
>    elsewhere, but I don't know where.  (Or my analysis of this being a problem
>    could have been completely wrong in the first place).  Leave the comment in
>    the code for so that someone may investigate it again in the future.
>    
>    Also add a bit of doxygen to transmit_provisional_response().
>    
>    (closes issue ASTERISK-18979)
>    
>    Review: https://reviewboard.asterisk.org/r/1648
>  ........
>................
>

Terry Wilson (351082):

>Don't prematurely stop SIP session timer
>
>When Asterisk is the UAS (incoming call, endpoint is re-inviting) the SIP session timer expires after half the time the sip endpoint indicates in the Session-expires header in proc_session_timer(). The session timer was being stopped totally and being handled as an error case instead of running again until the second expiry. This patch treats the half-time expiry as a non-error case and continues the timer until the true expiry.
>
>(closes issue ASTERISK-18996)
>Reported by: Thomas Arimont
>Tested by: Thomas Arimont
>Patches: session_timer_fix.diff by Terry Wilson (License #5357)
>  based on session_timer.patch by Thomas Arimont (License #5525)
>........
>
>Merged revisions 351080 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 351081 from http://svn.asterisk.org/svn/asterisk/branches/10
>


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Tests
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New Test Failures (1)
   - AsteriskTestSuite: S/channels/ s i p/codec negotiation
Existing Test Failures (1)
   - AsteriskTestSuite: S/fax/gateway mix2

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