[test-results] [Bamboo] Asterisk Testing - Asterisk Trunk - Asterisk CentOS 6 64-Bit 767 may have hung.
Bamboo
bamboo at asterisk.org
Tue Oct 30 19:25:03 CDT 2012
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TESTING-ASTERISKTRUNK-AST18CENTOS64-767 may have hung.
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This build has been running for 199 minutes, which is 149% 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-ASTERISKTRUNK-AST18CENTOS64/log
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Code Changes
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mmichelson (375442):
>Make evaluation of channel variables consistently case-sensitive.
>
>Due to inconsistencies in how variable names were evaluated, the
>decision was made to make all evaluations case-sensitive. See the
>UPGRADE.txt file or https://wiki.asterisk.org/wiki/display/AST/Case+Sensitivity
>for more details.
>
>(closes issue ASTERISK-20163)
>reported by Matt Jordan
>
>Review: https://reviewboard.asterisk.org/r/2160
>
mmichelson (375443):
>Prevent resetting of NATted realtime peer address on reload.
>
>If a "sip reload" is issued for a SIP peer, then his
>IP address will be cleared, thus resulting in forgetting the
>public IP address. Asterisk will then attempt to route SIP
>traffic to the private IP address.
>
>The fix here is to make "sip reload" ignore realtime peers
>when "host = dynamic" is spotted. Realtime peers can now only
>have their IP address reset if they have gone from being not
>dynamic to being dynamic.
>
>(closes issue ASTERISK-18203)
>reported by daren ferreira
>
>(closes issue ASTERISK-20572)
>reported by JoshE
>Patches:
> fix_nat_realtime.diff uploaded by JoshE (license #6075)
>........
>
>Merged revisions 375415 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 375417 from http://svn.asterisk.org/svn/asterisk/branches/10
>........
>
>Merged revisions 375437 from http://svn.asterisk.org/svn/asterisk/branches/11
>
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Last Logs
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30-Oct-2012 22:14:08 | Creating Asterisk instances ...
30-Oct-2012 22:14:08 | AMI 1 - connected, registering DTMF event...
30-Oct-2012 22:14:08 | AMI - originating call from SIP/testast1 to extension 1000 using extension 1000 at priority 1
30-Oct-2012 22:14:08 | AMI 2 - connected, registering DTMF event...
30-Oct-2012 22:14:08 | Received DTMF event from AMI 1...
30-Oct-2012 22:14:08 | Value of DTMF[digit] = 5
30-Oct-2012 22:14:50 | It's a match for ast[0] receiving DTMF from ast[1]
30-Oct-2012 22:14:50 | Received DTMF event from AMI 1...
30-Oct-2012 22:14:50 | Value of DTMF[digit] = 5
30-Oct-2012 22:14:50 | It's a match for ast[0] receiving DTMF from ast[1]
30-Oct-2012 22:14:50 | Received DTMF event from AMI 2...
30-Oct-2012 22:14:50 | Value of DTMF[digit] = 6
30-Oct-2012 22:14:50 | It's a match for ast[1] receiving DTMF from ast[0]
30-Oct-2012 22:14:50 | Both tones have been matched at least once. Test PASSED.
30-Oct-2012 22:14:50 | --> Running test 'tests/channels/SIP/sip_tls_register' ...
30-Oct-2012 22:14:50 |
30-Oct-2012 22:14:50 | Making sure Asterisk isn't running ...
30-Oct-2012 22:14:50 | Running ['tests/channels/SIP/sip_tls_register/run-test'] ...
30-Oct-2012 22:14:50 | --> Running test 'tests/channels/SIP/sip_srtp' ...
30-Oct-2012 22:14:50 |
30-Oct-2012 22:14:50 | Making sure Asterisk isn't running ...
30-Oct-2012 22:14:50 | Running ['tests/channels/SIP/sip_srtp/run-test'] ...
30-Oct-2012 22:14:50 | Initiating test call
30-Oct-2012 22:14:50 | Connection result 'SIP/2000-00000000 secure_media=1'
30-Oct-2012 22:14:50 | Connection result 'SIP/1000-00000000 secure_media=1'
30-Oct-2012 22:14:50 | Test passed
30-Oct-2012 22:14:50 | --> Running test 'tests/channels/SIP/noload_res_srtp' ...
30-Oct-2012 22:14:50 |
30-Oct-2012 22:14:50 | Making sure Asterisk isn't running ...
30-Oct-2012 22:14:50 | Running ['tests/channels/SIP/noload_res_srtp/run-test'] ...
30-Oct-2012 22:14:50 | Initiating test call
30-Oct-2012 22:14:50 | Connection result 'SIP/2000-00000000 secure_media='
30-Oct-2012 22:14:50 | Connection result 'SIP/1000-00000000 secure_media='
30-Oct-2012 22:14:50 | Test passed
30-Oct-2012 22:14:50 | self.connected_chan1: True
30-Oct-2012 22:14:50 | self.connected_no_srtp1:True
30-Oct-2012 22:14:50 | self.connected_chan2: True
30-Oct-2012 22:14:50 | self.connected_no_srtp2:True
30-Oct-2012 22:14:50 | --> Running test 'tests/channels/SIP/noload_res_srtp_attempt_srtp' ...
30-Oct-2012 22:14:50 |
30-Oct-2012 22:14:50 | Making sure Asterisk isn't running ...
30-Oct-2012 22:14:50 | Running ['tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test'] ...
30-Oct-2012 22:14:50 | AMI - connected
30-Oct-2012 22:14:50 | Initiating test call
30-Oct-2012 22:14:50 | Received VarSet event from AMI
30-Oct-2012 22:14:50 | Value of event[variable] = SIPCALLID
30-Oct-2012 22:14:50 | Value of event[value] = 5396059448a996407ad0373b46c296c2 at 127.0.0.1:5060
30-Oct-2012 22:14:50 | Received VarSet event from AMI
30-Oct-2012 22:14:50 | Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
30-Oct-2012 22:14:50 | Value of event[value] = SIP 401 Unauthorized
30-Oct-2012 22:14:50 | Received VarSet event from AMI
30-Oct-2012 22:14:50 | Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
30-Oct-2012 22:14:50 | Value of event[value] = SIP 401 Unauthorized
30-Oct-2012 22:14:50 | Received VarSet event from AMI
30-Oct-2012 22:14:50 | Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
30-Oct-2012 22:14:50 | Value of event[value] = SIP 488 Not acceptable here
30-Oct-2012 22:14:50 | self.connected_chan1:False
30-Oct-2012 22:14:50 | self.connected_srtp1:False
30-Oct-2012 22:14:50 | self.not_acceptable1:True
30-Oct-2012 22:14:50 | self.connected_chan2:False
30-Oct-2012 22:14:50 | self.connected_srtp2:False
30-Oct-2012 22:14:50 | Test passed
30-Oct-2012 22:14:50 | Received VarSet event from AMI
30-Oct-2012 22:14:50 | Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
30-Oct-2012 22:14:50 | Value of event[value] = SIP 488 Not acceptable here
30-Oct-2012 22:14:50 | Received VarSet event from AMI
30-Oct-2012 22:14:50 | Value of event[variable] = RTPAUDIOQOS
30-Oct-2012 22:14:50 | Value of event[value] = ssrc=1677740325;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000
30-Oct-2012 22:14:50 | Received VarSet event from AMI
30-Oct-2012 22:14:50 | Value of event[variable] = RTPAUDIOQOSJITTER
30-Oct-2012 22:14:50 | Value of event[value] = minrxjitter=0.000000;maxrxjitter=0.000000;avgrxjitter=0.000000;stdevrxjitter=0.000000;reported_minjitter=0.000000;reported_maxjitter=0.000000;reported_avgjitter=0.000000;reported_stdevjitter=0.000000;
30-Oct-2012 22:14:50 | Received VarSet event from AMI
30-Oct-2012 22:14:50 | Value of event[variable] = RTPAUDIOQOSLOSS
30-Oct-2012 22:14:50 | Value of event[value] = minrxlost=0.000000;maxrxlost=0.000000;avgrxlost=0.000000;stdevrxlost=0.000000;reported_minlost=0.000000;reported_maxlost=0.000000;reported_avglost=0.000000;reported_stdevlost=0.000000;
30-Oct-2012 22:14:50 | Received VarSet event from AMI
30-Oct-2012 22:14:50 | Value of event[variable] = RTPAUDIOQOSRTT
30-Oct-2012 22:14:50 | Value of event[value] = minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
30-Oct-2012 22:14:50 | Received VarSet event from AMI
30-Oct-2012 22:14:50 | Value of event[variable] = RTPAUDIOQOS
30-Oct-2012 22:14:50 | Value of event[value] = ssrc=1677740325;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000
30-Oct-2012 22:14:50 | self.connected_chan1:False
30-Oct-2012 22:14:50 | self.connected_srtp1:False
30-Oct-2012 22:14:50 | self.not_acceptable1:True
30-Oct-2012 22:14:50 | self.connected_chan2:False
30-Oct-2012 22:14:50 | self.connected_srtp2:False
30-Oct-2012 22:14:50 | Test passed
30-Oct-2012 22:14:50 | --> Running test 'tests/channels/SIP/secure_bridge_media' ...
30-Oct-2012 22:14:50 |
30-Oct-2012 22:14:50 | Making sure Asterisk isn't running ...
30-Oct-2012 22:14:50 | Running ['tests/channels/SIP/secure_bridge_media/run-test'] ...
30-Oct-2012 22:14:50 | Initiating test call
30-Oct-2012 22:14:50 | Connection result 'SIP/1000-00000000 secure_media=1'
30-Oct-2012 22:14:50 | Test passed
30-Oct-2012 22:14:50 | --> Running test 'tests/channels/SIP/message_disabled' ...
30-Oct-2012 22:14:50 |
30-Oct-2012 22:14:50 | Making sure Asterisk isn't running ...
30-Oct-2012 22:14:50 | Running ['tests/channels/SIP/message_disabled/run-test'] ...
30-Oct-2012 22:14:50 | --> Running test 'tests/channels/SIP/message_unauth' ...
30-Oct-2012 22:14:50 |
30-Oct-2012 22:14:50 | Making sure Asterisk isn't running ...
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