[test-results] [Bamboo] Asterisk Testing - Asterisk 11 Branch - Asterisk CentOS 6 64-Bit 197 may have hung.

Bamboo bamboo at asterisk.org
Mon Nov 19 19:00:40 CST 2012 

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TESTING-AST11BRANCH-AST18CENTOS64-197 may have hung.
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This build has been running for 202 minutes, which is 150% longer than usual.
It has been 187 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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wdoekes (376471):

>Fix most leftover non-opaque ast_str uses.
>
>Instead of calling str->str, one should use ast_str_buffer(str). Same
>goes for str->used as ast_str_strlen(str) and str->len as
>ast_str_size(str).
>
>Review: https://reviewboard.asterisk.org/r/2198
>........
>
>Merged revisions 376469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 376470 from http://svn.asterisk.org/svn/asterisk/branches/10
>



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Last Logs
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19-Nov-2012 21:43:48 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_refer_only/run-test'] ...
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/callee_with_reinvite' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/run-test'] ...
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_refer_only' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_refer_only/run-test'] ...
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_with_reinvite' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/run-test'] ...
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/sip_one_legged_transfer' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/sip_one_legged_transfer/run-test'] ...
19-Nov-2012 21:45:58 |  --> tests/channels/SIP/sip_one_legged_transfer_v6 ... skipped 'Skip while failures are debugged'
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/sip_register' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/sip_register/run-test'] ...
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/sip_register_domain_acl' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/sip_register_domain_acl/run-test'] ...
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/sip_channel_params' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/sip_channel_params/run-test'] ...
19-Nov-2012 21:45:58 |  Got channel name SIP/test1-00000000
19-Nov-2012 21:45:58 |  test passed
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/sip_tls_call' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/sip_tls_call/run-test'] ...
19-Nov-2012 21:45:58 |  Building test resources ...
19-Nov-2012 21:45:58 |  Creating Asterisk instances ...
19-Nov-2012 21:45:58 |  AMI 1 - connected, registering DTMF event...
19-Nov-2012 21:45:58 |  AMI - originating call from SIP/testast1 to extension 1000 using extension 1000 at priority 1
19-Nov-2012 21:45:58 |  AMI 2 - connected, registering DTMF event...
19-Nov-2012 21:45:58 |  Received DTMF event from AMI 1...
19-Nov-2012 21:45:58 |  Value of DTMF[digit] = 5
19-Nov-2012 21:45:58 |  It's a match for ast[0] receiving DTMF from ast[1]
19-Nov-2012 21:45:58 |  Received DTMF event from AMI 1...
19-Nov-2012 21:45:58 |  Value of DTMF[digit] = 5
19-Nov-2012 21:45:58 |  It's a match for ast[0] receiving DTMF from ast[1]
19-Nov-2012 21:45:58 |  Received DTMF event from AMI 2...
19-Nov-2012 21:45:58 |  Value of DTMF[digit] = 6
19-Nov-2012 21:45:58 |  It's a match for ast[1] receiving DTMF from ast[0]
19-Nov-2012 21:45:58 |  Both tones have been matched at least once.  Test PASSED.
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/sip_tls_register' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/sip_tls_register/run-test'] ...
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/sip_srtp' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/sip_srtp/run-test'] ...
19-Nov-2012 21:45:58 |  Initiating test call
19-Nov-2012 21:45:58 |  Connection result 'SIP/2000-00000000 secure_media=1'
19-Nov-2012 21:45:58 |  Connection result 'SIP/1000-00000000 secure_media=1'
19-Nov-2012 21:45:58 |  Test passed
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/noload_res_srtp' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/noload_res_srtp/run-test'] ...
19-Nov-2012 21:45:58 |  Initiating test call
19-Nov-2012 21:45:58 |  Connection result 'SIP/2000-00000000 secure_media='
19-Nov-2012 21:45:58 |  Connection result 'SIP/1000-00000000 secure_media='
19-Nov-2012 21:45:58 |  Test passed
19-Nov-2012 21:45:58 |  self.connected_chan1:   True
19-Nov-2012 21:45:58 |  self.connected_no_srtp1:True
19-Nov-2012 21:45:58 |  self.connected_chan2:   True
19-Nov-2012 21:45:58 |  self.connected_no_srtp2:True
19-Nov-2012 21:45:58 |  --> Running test 'tests/channels/SIP/noload_res_srtp_attempt_srtp' ...
19-Nov-2012 21:45:58 |  
19-Nov-2012 21:45:58 |  Making sure Asterisk isn't running ...
19-Nov-2012 21:45:58 |  Running ['tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test'] ...
19-Nov-2012 21:45:58 |  AMI - connected
19-Nov-2012 21:45:58 |  Initiating test call
19-Nov-2012 21:45:58 |  Received VarSet event from AMI
19-Nov-2012 21:45:58 |    Value of event[variable] = SIPCALLID
19-Nov-2012 21:45:58 |    Value of event[value] = 3c24ac88617eef380ead548c7a260ca3 at 127.0.0.1:5060
19-Nov-2012 21:45:58 |  Received VarSet event from AMI
19-Nov-2012 21:45:58 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
19-Nov-2012 21:45:58 |    Value of event[value] = SIP 401 Unauthorized
19-Nov-2012 21:45:58 |  Received VarSet event from AMI
19-Nov-2012 21:45:58 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
19-Nov-2012 21:45:58 |    Value of event[value] = SIP 401 Unauthorized
19-Nov-2012 21:45:58 |  Received VarSet event from AMI
19-Nov-2012 21:45:58 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
19-Nov-2012 21:45:58 |    Value of event[value] = SIP 488 Not acceptable here
19-Nov-2012 21:45:58 |  self.connected_chan1:False
19-Nov-2012 21:45:58 |  self.connected_srtp1:False
19-Nov-2012 21:45:58 |  self.not_acceptable1:True
19-Nov-2012 21:45:58 |  self.connected_chan2:False
19-Nov-2012 21:45:58 |  self.connected_srtp2:False
19-Nov-2012 21:45:58 |  Test passed

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