[test-results] [Bamboo] Asterisk Testing - Asterisk 11 Branch - Asterisk CentOS 6 32-Bit 484 may have hung.

[Bamboo]

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TESTING-AST11BRANCH-AST18CENTOS32-484 may have hung.
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This build has been running for 192 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-32-centos.digium.internal

http://bamboo.asterisk.org/browse/TESTING-AST11BRANCH-AST18CENTOS32/log

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Code Changes
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mjordan (391241):

>Add announce-to-first-user option for app_queue
>
>In r386792, the ability to play prompts to the first caller in a call queue was
>added. While this is arguably a bug fix for those who expect the first caller
>to continue receiving prompts while the agent is dialed, it has the side effect
>of preventing the first caller from hearing the agent immediately upon
>bridging. This may not be a problem for those who really want this option, but
>for those who didn't care whether or not the first caller in queue heard their
>position, it was an issue.
>
>This patch disables the ability for the first caller in the queue to hear
>prompts and adds a new option, announce-to-first-user, to queues.conf. Those
>who the behavior can enable it by setting this value to True.
>
>Note that if we ever implement the ability to have the prompts be stopped
>upon bridging, this option can be removed.
>
>(closes issue ASTERISK-21782)
>Reported by: Remi Quezada
>........
>
>Merged revisions 391215 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>



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Last Logs
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12-Jun-2013 16:26:54 |  --- --> Tags: ['SIP', 'transfer']
12-Jun-2013 16:26:54 |  --- --> Dependency: twisted - True
12-Jun-2013 16:26:54 |  --- --> Dependency: starpy - True
12-Jun-2013 16:26:54 |  --- --> Dependency: pjsua - False
12-Jun-2013 16:26:54 |  
12-Jun-2013 16:28:29 |  --> tests/channels/SIP/sip_attended_transfer_tcp ... skipped 'See ASTERISK-20614'
12-Jun-2013 16:28:29 |  --> tests/channels/SIP/sip_attended_transfer_v6 ... skipped 'See ASTERISK-20616'
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/callee_refer_only' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_refer_only/run-test'] ...
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/callee_with_reinvite' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/run-test'] ...
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_refer_only' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_refer_only/run-test'] ...
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_with_reinvite' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/run-test'] ...
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/sip_one_legged_transfer' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/sip_one_legged_transfer/run-test'] ...
12-Jun-2013 16:28:29 |  [Jun 12 17:27:32] WARNING[13130]: __main__:64 checkBridgeResult: 'link' and 'bridgedchannel' not found
12-Jun-2013 16:28:29 |  --> tests/channels/SIP/sip_one_legged_transfer_v6 ... skipped 'Skip while failures are debugged'
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/sip_register' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/sip_register/run-test'] ...
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/sip_register_domain_acl' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/sip_register_domain_acl/run-test'] ...
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/sip_channel_params' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/sip_channel_params/run-test'] ...
12-Jun-2013 16:28:29 |  Got channel name SIP/test1-00000000
12-Jun-2013 16:28:29 |  test passed
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/sip_tls_call' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/sip_tls_call/run-test'] ...
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/sip_tls_register' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/sip_tls_register/run-test'] ...
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/sip_srtp' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/sip_srtp/run-test'] ...
12-Jun-2013 16:28:29 |  Initiating test call
12-Jun-2013 16:28:29 |  Connection result 'SIP/2000-00000000 secure_media=1'
12-Jun-2013 16:28:29 |  Connection result 'SIP/1000-00000000 secure_media=1'
12-Jun-2013 16:28:29 |  Test passed
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/noload_res_srtp' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/noload_res_srtp/run-test'] ...
12-Jun-2013 16:28:29 |  Initiating test call
12-Jun-2013 16:28:29 |  Connection result 'SIP/2000-00000000 secure_media='
12-Jun-2013 16:28:29 |  Connection result 'SIP/1000-00000000 secure_media='
12-Jun-2013 16:28:29 |  Test passed
12-Jun-2013 16:28:29 |  self.connected_chan1:   True
12-Jun-2013 16:28:29 |  self.connected_no_srtp1:True
12-Jun-2013 16:28:29 |  self.connected_chan2:   True
12-Jun-2013 16:28:29 |  self.connected_no_srtp2:True
12-Jun-2013 16:28:29 |  --> Running test 'tests/channels/SIP/noload_res_srtp_attempt_srtp' ...
12-Jun-2013 16:28:29 |  
12-Jun-2013 16:28:29 |  Making sure Asterisk isn't running ...
12-Jun-2013 16:28:29 |  Running ['tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test'] ...
12-Jun-2013 16:28:29 |  AMI - connected
12-Jun-2013 16:28:29 |  Initiating test call
12-Jun-2013 16:28:29 |  Received VarSet event from AMI
12-Jun-2013 16:28:29 |    Value of event[variable] = SIPCALLID
12-Jun-2013 16:28:29 |    Value of event[value] = 6377fe7869b57b040a0f8d92402fd5c5 at 127.0.0.1:5060
12-Jun-2013 16:28:29 |  Received VarSet event from AMI
12-Jun-2013 16:28:29 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
12-Jun-2013 16:28:29 |    Value of event[value] = SIP 401 Unauthorized
12-Jun-2013 16:28:29 |  Received VarSet event from AMI
12-Jun-2013 16:28:29 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
12-Jun-2013 16:28:29 |    Value of event[value] = SIP 401 Unauthorized
12-Jun-2013 16:28:29 |  Received VarSet event from AMI
12-Jun-2013 16:28:29 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
12-Jun-2013 16:28:29 |    Value of event[value] = SIP 488 Not acceptable here
12-Jun-2013 16:28:29 |  self.connected_chan1:False
12-Jun-2013 16:28:29 |  self.connected_srtp1:False
12-Jun-2013 16:28:29 |  self.not_acceptable1:True
12-Jun-2013 16:28:29 |  self.connected_chan2:False
12-Jun-2013 16:28:29 |  self.connected_srtp2:False
12-Jun-2013 16:28:29 |  Test passed
12-Jun-2013 16:28:29 |  Received VarSet event from AMI
12-Jun-2013 16:28:29 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
12-Jun-2013 16:28:29 |    Value of event[value] = SIP 488 Not acceptable here
12-Jun-2013 16:28:29 |  Received VarSet event from AMI
12-Jun-2013 16:28:29 |    Value of event[variable] = RTPAUDIOQOS

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