[test-results] [Bamboo] Asterisk Testing - Asterisk 11 Branch - Asterisk CentOS 6 64-Bit 116 may have hung.

[Bamboo]

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TESTING-AST11BRANCH-AST18CENTOS64-116 may have hung.
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This build has been running for 198 minutes, which is 150% longer than usual.
It has been 183 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-AST11BRANCH-AST18CENTOS64/log

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Code Changes
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mmichelson (374106):

>Don't destroy confbridge config when error is encountered during a reload.
>
>Not panicking means that the old config is kept.
>
>(closes issue ASTERISK-20458)
>Reported by: Leif Madsen
>Patches:
>	ASTERISK-20458.patch uploaded by Mark Michelson(license #5049)
>Tested by Leif Madsen
>
>



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Last Logs
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01-Oct-2012 17:21:47 |  --> Running test 'tests/channels/SIP/sip_attended_transfer_tcp' ...
01-Oct-2012 17:21:47 |  
01-Oct-2012 17:21:47 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:21:47 |  Running ['tests/channels/SIP/sip_attended_transfer_tcp/run-test'] ...
01-Oct-2012 17:23:56 |  --> tests/channels/SIP/sip_attended_transfer_v6 ... skipped 'Skip while failures are debugged'
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/callee_refer_only' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_refer_only/run-test'] ...
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/callee_with_reinvite' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/run-test'] ...
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_refer_only' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_refer_only/run-test'] ...
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_with_reinvite' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/run-test'] ...
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/sip_one_legged_transfer' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/sip_one_legged_transfer/run-test'] ...
01-Oct-2012 17:23:56 |  --> tests/channels/SIP/sip_one_legged_transfer_v6 ... skipped 'Skip while failures are debugged'
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/sip_register' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/sip_register/run-test'] ...
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/sip_register_domain_acl' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/sip_register_domain_acl/run-test'] ...
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/sip_channel_params' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/sip_channel_params/run-test'] ...
01-Oct-2012 17:23:56 |  Got channel name SIP/test1-00000000
01-Oct-2012 17:23:56 |  test passed
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/sip_tls_call' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/sip_tls_call/run-test'] ...
01-Oct-2012 17:23:56 |  Building test resources ...
01-Oct-2012 17:23:56 |  Creating Asterisk instances ...
01-Oct-2012 17:23:56 |  AMI 1 - connected, registering DTMF event...
01-Oct-2012 17:23:56 |  AMI - originating call from SIP/testast1 to extension 1000 using extension 1000 at priority 1
01-Oct-2012 17:23:56 |  AMI 2 - connected, registering DTMF event...
01-Oct-2012 17:23:56 |  Received DTMF event from AMI 1...
01-Oct-2012 17:23:56 |  Value of DTMF[digit] = 5
01-Oct-2012 17:23:56 |  It's a match for ast[0] receiving DTMF from ast[1]
01-Oct-2012 17:23:56 |  Received DTMF event from AMI 1...
01-Oct-2012 17:23:56 |  Value of DTMF[digit] = 5
01-Oct-2012 17:23:56 |  It's a match for ast[0] receiving DTMF from ast[1]
01-Oct-2012 17:23:56 |  Received DTMF event from AMI 2...
01-Oct-2012 17:23:56 |  Value of DTMF[digit] = 6
01-Oct-2012 17:23:56 |  It's a match for ast[1] receiving DTMF from ast[0]
01-Oct-2012 17:23:56 |  Both tones have been matched at least once.  Test PASSED.
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/sip_tls_register' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/sip_tls_register/run-test'] ...
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/sip_srtp' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/sip_srtp/run-test'] ...
01-Oct-2012 17:23:56 |  Initiating test call
01-Oct-2012 17:23:56 |  Connection result 'SIP/2000-00000000 secure_media=1'
01-Oct-2012 17:23:56 |  Connection result 'SIP/1000-00000000 secure_media=1'
01-Oct-2012 17:23:56 |  Test passed
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/noload_res_srtp' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/noload_res_srtp/run-test'] ...
01-Oct-2012 17:23:56 |  Initiating test call
01-Oct-2012 17:23:56 |  Connection result 'SIP/2000-00000000 secure_media='
01-Oct-2012 17:23:56 |  Connection result 'SIP/1000-00000000 secure_media='
01-Oct-2012 17:23:56 |  Test passed
01-Oct-2012 17:23:56 |  self.connected_chan1:   True
01-Oct-2012 17:23:56 |  self.connected_no_srtp1:True
01-Oct-2012 17:23:56 |  self.connected_chan2:   True
01-Oct-2012 17:23:56 |  self.connected_no_srtp2:True
01-Oct-2012 17:23:56 |  --> Running test 'tests/channels/SIP/noload_res_srtp_attempt_srtp' ...
01-Oct-2012 17:23:56 |  
01-Oct-2012 17:23:56 |  Making sure Asterisk isn't running ...
01-Oct-2012 17:23:56 |  Running ['tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test'] ...
01-Oct-2012 17:23:56 |  AMI - connected
01-Oct-2012 17:23:56 |  Initiating test call
01-Oct-2012 17:23:56 |  Received VarSet event from AMI
01-Oct-2012 17:23:56 |    Value of event[variable] = SIPCALLID
01-Oct-2012 17:23:56 |    Value of event[value] = 2cccdbb737e8115f268026f579348a68 at 127.0.0.1:5060
01-Oct-2012 17:23:56 |  Received VarSet event from AMI
01-Oct-2012 17:23:56 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
01-Oct-2012 17:23:56 |    Value of event[value] = SIP 401 Unauthorized
01-Oct-2012 17:23:56 |  Received VarSet event from AMI
01-Oct-2012 17:23:56 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
01-Oct-2012 17:23:56 |    Value of event[value] = SIP 401 Unauthorized
01-Oct-2012 17:23:56 |  Received VarSet event from AMI
01-Oct-2012 17:23:56 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~

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