[test-results] [Bamboo] Asterisk Testing - Asterisk 10 Branch - Asterisk CentOS 6 64-Bit 410 may have hung.

[Bamboo]

-------------------------------------------------------------------------------
TESTING-ASTERISK10BRANCH-AST18CENTOS64-410 may have hung.
-------------------------------------------------------------------------------

This build has been running for 179 minutes, which is 150% longer than usual.
It has been 166 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-ASTERISK10BRANCH-AST18CENTOS64/log

--------------
Code Changes
--------------
seanbright (374132):

>Use ast_copy_string instead of strncpy to guarantee a NUL terminated string.
>



--------------
Last Logs
--------------
02-Oct-2012 04:56:46 |  
02-Oct-2012 04:56:46 |  --> Running test 'tests/channels/SIP/sip_attended_transfer_tcp' ...
02-Oct-2012 04:56:46 |  
02-Oct-2012 04:56:46 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_attended_transfer_tcp/run-test'] ...
02-Oct-2012 04:58:54 |  --> tests/channels/SIP/sip_attended_transfer_v6 ... skipped 'Skip while failures are debugged'
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/callee_refer_only' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_refer_only/run-test'] ...
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/callee_with_reinvite' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/run-test'] ...
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_refer_only' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_refer_only/run-test'] ...
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_with_reinvite' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/run-test'] ...
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/sip_one_legged_transfer' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_one_legged_transfer/run-test'] ...
02-Oct-2012 04:58:54 |  --> tests/channels/SIP/sip_one_legged_transfer_v6 ... skipped 'Skip while failures are debugged'
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/sip_register' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_register/run-test'] ...
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/sip_register_domain_acl' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_register_domain_acl/run-test'] ...
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/sip_channel_params' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_channel_params/run-test'] ...
02-Oct-2012 04:58:54 |  Got channel name SIP/test1-00000000
02-Oct-2012 04:58:54 |  test passed
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/sip_tls_call' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_tls_call/run-test'] ...
02-Oct-2012 04:58:54 |  Building test resources ...
02-Oct-2012 04:58:54 |  Creating Asterisk instances ...
02-Oct-2012 04:58:54 |  AMI 2 - connected, registering DTMF event...
02-Oct-2012 04:58:54 |  AMI 1 - connected, registering DTMF event...
02-Oct-2012 04:58:54 |  AMI - originating call from SIP/testast1 to extension 1000 using extension 1000 at priority 1
02-Oct-2012 04:58:54 |  Received DTMF event from AMI 1...
02-Oct-2012 04:58:54 |  Value of DTMF[digit] = 5
02-Oct-2012 04:58:54 |  It's a match for ast[0] receiving DTMF from ast[1]
02-Oct-2012 04:58:54 |  Received DTMF event from AMI 1...
02-Oct-2012 04:58:54 |  Value of DTMF[digit] = 5
02-Oct-2012 04:58:54 |  It's a match for ast[0] receiving DTMF from ast[1]
02-Oct-2012 04:58:54 |  Received DTMF event from AMI 2...
02-Oct-2012 04:58:54 |  Value of DTMF[digit] = 6
02-Oct-2012 04:58:54 |  It's a match for ast[1] receiving DTMF from ast[0]
02-Oct-2012 04:58:54 |  Both tones have been matched at least once.  Test PASSED.
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/sip_tls_register' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_tls_register/run-test'] ...
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/sip_srtp' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/sip_srtp/run-test'] ...
02-Oct-2012 04:58:54 |  Initiating test call
02-Oct-2012 04:58:54 |  Connection result 'SIP/2000-00000000 secure_media=1'
02-Oct-2012 04:58:54 |  Connection result 'SIP/1000-00000000 secure_media=1'
02-Oct-2012 04:58:54 |  Test passed
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/noload_res_srtp' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/noload_res_srtp/run-test'] ...
02-Oct-2012 04:58:54 |  Initiating test call
02-Oct-2012 04:58:54 |  Connection result 'SIP/2000-00000000 secure_media='
02-Oct-2012 04:58:54 |  Connection result 'SIP/1000-00000000 secure_media='
02-Oct-2012 04:58:54 |  Test passed
02-Oct-2012 04:58:54 |  self.connected_chan1:   True
02-Oct-2012 04:58:54 |  self.connected_no_srtp1:True
02-Oct-2012 04:58:54 |  self.connected_chan2:   True
02-Oct-2012 04:58:54 |  self.connected_no_srtp2:True
02-Oct-2012 04:58:54 |  --> Running test 'tests/channels/SIP/noload_res_srtp_attempt_srtp' ...
02-Oct-2012 04:58:54 |  
02-Oct-2012 04:58:54 |  Making sure Asterisk isn't running ...
02-Oct-2012 04:58:54 |  Running ['tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test'] ...
02-Oct-2012 04:58:54 |  AMI - connected
02-Oct-2012 04:58:54 |  Initiating test call
02-Oct-2012 04:58:54 |  Received VarSet event from AMI
02-Oct-2012 04:58:54 |    Value of event[variable] = SIPCALLID
02-Oct-2012 04:58:54 |    Value of event[value] = 266f0fe1110926e25cc6d8ff10280154 at 127.0.0.1:5060
02-Oct-2012 04:58:54 |  Received VarSet event from AMI
02-Oct-2012 04:58:54 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
02-Oct-2012 04:58:54 |    Value of event[value] = SIP 401 Unauthorized
02-Oct-2012 04:58:54 |  Received VarSet event from AMI
02-Oct-2012 04:58:54 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
02-Oct-2012 04:58:54 |    Value of event[value] = SIP 488 Not acceptable here
02-Oct-2012 04:58:54 |  self.connected_chan1:False

--
This message is automatically generated by Atlassian Bamboo
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/test-results/attachments/20121002/6faa7ade/attachment-0001.htm>