[test-results] [Bamboo] Asterisk Testing - Asterisk Trunk - Asterisk CentOS 6 64-Bit 797 may have hung.

[Bamboo]

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TESTING-ASTERISKTRUNK-AST18CENTOS64-797 may have hung.
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This build has been running for 203 minutes, which is 149% longer than usual.
It has been 189 minutes since Bamboo received a log message for this build.
Running on agent asterisk-testsuite-01.digium.internal.digium.internal

http://bamboo.asterisk.org/browse/TESTING-ASTERISKTRUNK-AST18CENTOS64/log

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Code Changes
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jrose (376312):

>app_meetme: Fix channels lingering when hung up under certain conditions
>
>Channels would get stuck and MeetMe would repeatedly display an Unable
>to write frame to channel error in the conf_run function if hung up
>during certain sound prompts such as during user count announcements.
>This patch fixes that by reintroducing a hangup check in the meetme's
>main loop (also in conf_run).
>
>(closes issue ASTERISK-20486)
>Reported by: Michael Cargile
>Review: https://reviewboard.asterisk.org/r/2187/
>Patches:
>    meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan Rose (license 6182)
>........
>
>Merged revisions 376307 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 376308 from http://svn.asterisk.org/svn/asterisk/branches/10
>........
>
>Merged revisions 376310 from http://svn.asterisk.org/svn/asterisk/branches/11
>



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Last Logs
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16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_refer_only/run-test'] ...
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/callee_with_reinvite' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/run-test'] ...
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_refer_only' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_refer_only/run-test'] ...
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_with_reinvite' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/run-test'] ...
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/sip_one_legged_transfer' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/sip_one_legged_transfer/run-test'] ...
16-Nov-2012 01:55:26 |  --> tests/channels/SIP/sip_one_legged_transfer_v6 ... skipped 'Skip while failures are debugged'
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/sip_register' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/sip_register/run-test'] ...
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/sip_register_domain_acl' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/sip_register_domain_acl/run-test'] ...
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/sip_channel_params' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/sip_channel_params/run-test'] ...
16-Nov-2012 01:55:26 |  Got channel name SIP/test1-00000000
16-Nov-2012 01:55:26 |  test passed
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/sip_tls_call' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/sip_tls_call/run-test'] ...
16-Nov-2012 01:55:26 |  Building test resources ...
16-Nov-2012 01:55:26 |  Creating Asterisk instances ...
16-Nov-2012 01:55:26 |  AMI 1 - connected, registering DTMF event...
16-Nov-2012 01:55:26 |  AMI - originating call from SIP/testast1 to extension 1000 using extension 1000 at priority 1
16-Nov-2012 01:55:26 |  AMI 2 - connected, registering DTMF event...
16-Nov-2012 01:55:26 |  Received DTMF event from AMI 1...
16-Nov-2012 01:55:26 |  Value of DTMF[digit] = 5
16-Nov-2012 01:55:26 |  It's a match for ast[0] receiving DTMF from ast[1]
16-Nov-2012 01:55:26 |  Received DTMF event from AMI 1...
16-Nov-2012 01:55:26 |  Value of DTMF[digit] = 5
16-Nov-2012 01:55:26 |  It's a match for ast[0] receiving DTMF from ast[1]
16-Nov-2012 01:55:26 |  Received DTMF event from AMI 2...
16-Nov-2012 01:55:26 |  Value of DTMF[digit] = 6
16-Nov-2012 01:55:26 |  It's a match for ast[1] receiving DTMF from ast[0]
16-Nov-2012 01:55:26 |  Both tones have been matched at least once.  Test PASSED.
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/sip_tls_register' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/sip_tls_register/run-test'] ...
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/sip_srtp' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/sip_srtp/run-test'] ...
16-Nov-2012 01:55:26 |  Initiating test call
16-Nov-2012 01:55:26 |  Connection result 'SIP/2000-00000000 secure_media=1'
16-Nov-2012 01:55:26 |  Connection result 'SIP/1000-00000000 secure_media=1'
16-Nov-2012 01:55:26 |  Test passed
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/noload_res_srtp' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/noload_res_srtp/run-test'] ...
16-Nov-2012 01:55:26 |  Initiating test call
16-Nov-2012 01:55:26 |  Connection result 'SIP/2000-00000000 secure_media='
16-Nov-2012 01:55:26 |  Connection result 'SIP/1000-00000000 secure_media='
16-Nov-2012 01:55:26 |  Test passed
16-Nov-2012 01:55:26 |  self.connected_chan1:   True
16-Nov-2012 01:55:26 |  self.connected_no_srtp1:True
16-Nov-2012 01:55:26 |  self.connected_chan2:   True
16-Nov-2012 01:55:26 |  self.connected_no_srtp2:True
16-Nov-2012 01:55:26 |  --> Running test 'tests/channels/SIP/noload_res_srtp_attempt_srtp' ...
16-Nov-2012 01:55:26 |  
16-Nov-2012 01:55:26 |  Making sure Asterisk isn't running ...
16-Nov-2012 01:55:26 |  Running ['tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test'] ...
16-Nov-2012 01:55:26 |  AMI - connected
16-Nov-2012 01:55:26 |  Initiating test call
16-Nov-2012 01:55:26 |  Received VarSet event from AMI
16-Nov-2012 01:55:26 |    Value of event[variable] = SIPCALLID
16-Nov-2012 01:55:26 |    Value of event[value] = 107c0225682d9f3256a9c654245734d0 at 127.0.0.1:5060
16-Nov-2012 01:55:26 |  Received VarSet event from AMI
16-Nov-2012 01:55:26 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
16-Nov-2012 01:55:26 |    Value of event[value] = SIP 401 Unauthorized
16-Nov-2012 01:55:26 |  Received VarSet event from AMI
16-Nov-2012 01:55:26 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
16-Nov-2012 01:55:26 |    Value of event[value] = SIP 401 Unauthorized
16-Nov-2012 01:55:26 |  Received VarSet event from AMI
16-Nov-2012 01:55:26 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
16-Nov-2012 01:55:26 |    Value of event[value] = SIP 488 Not acceptable here
16-Nov-2012 01:55:26 |  self.connected_chan1:False
16-Nov-2012 01:55:26 |  self.connected_srtp1:False
16-Nov-2012 01:55:26 |  self.not_acceptable1:True
16-Nov-2012 01:55:26 |  self.connected_chan2:False
16-Nov-2012 01:55:26 |  self.connected_srtp2:False
16-Nov-2012 01:55:26 |  Test passed
16-Nov-2012 01:55:26 |  Received VarSet event from AMI

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