[test-results] [Bamboo] Asterisk Testing - Asterisk Trunk - Asterisk CentOS 6 64-Bit 385 may have hung.

Bamboo bamboo at asterisk.org
Wed Jun 20 18:43:44 CDT 2012


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TESTING-ASTERISKTRUNK-AST18CENTOS64-385 may have hung.
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This build has been running for 178 minutes, which is 150% longer than usual.
It has been 165 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
--------------
Mark Michelson (369068):

>Fix request routing issue when outboundproxy is used.
>
>Asterisk was incorrectly setting the destination of CANCELs
>and ACKs for error responses to the URI of the initial INVITE.
>This resulted in further requests, such as INVITEs with authentication
>credentials, to be routed incorrectly. Instead, when these CANCEL
>or ACKs are to be sent, we should simply keep the destination the
>same as what it previously was. There is no need to alter it any.
>
>(closes issue ASTERISK-20008)
>Reported by Marcus Hunger
>Patches:
>	ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
>........
>
>Merged revisions 369066 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 369067 from http://svn.asterisk.org/svn/asterisk/branches/10
>



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Last Logs
--------------
20-Jun-2012 21:42:40 |  --> Running test 'tests/channels/SIP/sip_channel_params' ...
20-Jun-2012 21:42:40 |  
20-Jun-2012 21:42:40 |  Making sure Asterisk isn't running ...
20-Jun-2012 21:42:40 |  Running ['tests/channels/SIP/sip_channel_params/run-test'] ...
20-Jun-2012 21:42:40 |  Got channel name SIP/test1-00000000
20-Jun-2012 21:42:40 |  test passed
20-Jun-2012 21:42:40 |  --> Running test 'tests/channels/SIP/sip_tls_call' ...
20-Jun-2012 21:42:40 |  
20-Jun-2012 21:42:40 |  Making sure Asterisk isn't running ...
20-Jun-2012 21:43:17 |  Running ['tests/channels/SIP/sip_tls_call/run-test'] ...
20-Jun-2012 21:43:17 |  Building test resources ...
20-Jun-2012 21:43:17 |  Creating Asterisk instances ...
20-Jun-2012 21:43:17 |  AMI 1 - connected, registering DTMF event...
20-Jun-2012 21:43:17 |  AMI - originating call from SIP/testast1 to extension 1000 using extension 1000 at priority 1
20-Jun-2012 21:43:17 |  AMI 2 - connected, registering DTMF event...
20-Jun-2012 21:43:17 |  Received DTMF event from AMI 1...
20-Jun-2012 21:43:17 |  Value of DTMF[digit] = 5
20-Jun-2012 21:43:17 |  It's a match for ast[0] receiving DTMF from ast[1]
20-Jun-2012 21:43:17 |  Received DTMF event from AMI 1...
20-Jun-2012 21:43:17 |  Value of DTMF[digit] = 5
20-Jun-2012 21:43:17 |  It's a match for ast[0] receiving DTMF from ast[1]
20-Jun-2012 21:43:17 |  Received DTMF event from AMI 2...
20-Jun-2012 21:43:17 |  Value of DTMF[digit] = 6
20-Jun-2012 21:43:17 |  It's a match for ast[1] receiving DTMF from ast[0]
20-Jun-2012 21:43:17 |  Both tones have been matched at least once.  Test PASSED.
20-Jun-2012 21:43:17 |  --> Running test 'tests/channels/SIP/sip_tls_register' ...
20-Jun-2012 21:43:17 |  
20-Jun-2012 21:43:17 |  Making sure Asterisk isn't running ...
20-Jun-2012 21:43:17 |  Running ['tests/channels/SIP/sip_tls_register/run-test'] ...
20-Jun-2012 21:43:17 |  --> Running test 'tests/channels/SIP/sip_srtp' ...
20-Jun-2012 21:43:17 |  
20-Jun-2012 21:43:17 |  Making sure Asterisk isn't running ...
20-Jun-2012 21:43:17 |  Running ['tests/channels/SIP/sip_srtp/run-test'] ...
20-Jun-2012 21:43:17 |  Initiating test call
20-Jun-2012 21:43:17 |  Connection result 'SIP/2000-00000000 secure_media=1'
20-Jun-2012 21:43:17 |  Connection result 'SIP/1000-00000000 secure_media=1'
20-Jun-2012 21:43:17 |  Test passed
20-Jun-2012 21:43:17 |  --> Running test 'tests/channels/SIP/noload_res_srtp' ...
20-Jun-2012 21:43:17 |  
20-Jun-2012 21:43:17 |  Making sure Asterisk isn't running ...
20-Jun-2012 21:43:17 |  Running ['tests/channels/SIP/noload_res_srtp/run-test'] ...
20-Jun-2012 21:43:17 |  Initiating test call
20-Jun-2012 21:43:17 |  Connection result 'SIP/2000-00000000 secure_media='
20-Jun-2012 21:43:17 |  Connection result 'SIP/1000-00000000 secure_media='
20-Jun-2012 21:43:17 |  Test passed
20-Jun-2012 21:43:17 |  self.connected_chan1:   True
20-Jun-2012 21:43:17 |  self.connected_no_srtp1:True
20-Jun-2012 21:43:17 |  self.connected_chan2:   True
20-Jun-2012 21:43:17 |  self.connected_no_srtp2:True
20-Jun-2012 21:43:17 |  --> Running test 'tests/channels/SIP/noload_res_srtp_attempt_srtp' ...
20-Jun-2012 21:43:17 |  
20-Jun-2012 21:43:17 |  Making sure Asterisk isn't running ...
20-Jun-2012 21:43:17 |  Running ['tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test'] ...
20-Jun-2012 21:43:17 |  AMI - connected
20-Jun-2012 21:43:17 |  Initiating test call
20-Jun-2012 21:43:17 |  Received VarSet event from AMI
20-Jun-2012 21:43:17 |    Value of event[variable] = SIPCALLID
20-Jun-2012 21:43:17 |    Value of event[value] = 006ab0ee4e8f860a13ffaf8c6053a51e at 127.0.0.1:5060
20-Jun-2012 21:43:17 |  Received VarSet event from AMI
20-Jun-2012 21:43:17 |    Value of event[variable] = ~HASH~HANGUPCAUSE~SIP/2000-00000000~
20-Jun-2012 21:43:17 |    Value of event[value] = SIP 401 Unauthorized
20-Jun-2012 21:43:17 |  Received VarSet event from AMI
20-Jun-2012 21:43:17 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
20-Jun-2012 21:43:17 |    Value of event[value] = SIP 401 Unauthorized
20-Jun-2012 21:43:17 |  Received VarSet event from AMI
20-Jun-2012 21:43:17 |    Value of event[variable] = ~HASH~HANGUPCAUSE~SIP/2000-00000000~
20-Jun-2012 21:43:17 |    Value of event[value] = SIP 488 Not acceptable here
20-Jun-2012 21:43:17 |  Received VarSet event from AMI
20-Jun-2012 21:43:17 |    Value of event[variable] = ~HASH~SIP_CAUSE~SIP/2000-00000000~
20-Jun-2012 21:43:17 |    Value of event[value] = SIP 488 Not acceptable here
20-Jun-2012 21:43:17 |  self.connected_chan1:False
20-Jun-2012 21:43:17 |  self.connected_srtp1:False
20-Jun-2012 21:43:17 |  self.not_acceptable1:True
20-Jun-2012 21:43:17 |  self.connected_chan2:False
20-Jun-2012 21:43:17 |  self.connected_srtp2:False
20-Jun-2012 21:43:17 |  Test passed
20-Jun-2012 21:43:17 |  Received VarSet event from AMI
20-Jun-2012 21:43:17 |    Value of event[variable] = RTPAUDIOQOS
20-Jun-2012 21:43:17 |    Value of event[value] = ssrc=1268035130;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000
20-Jun-2012 21:43:17 |  Received VarSet event from AMI
20-Jun-2012 21:43:17 |    Value of event[variable] = RTPAUDIOQOSJITTER
20-Jun-2012 21:43:17 |    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;
20-Jun-2012 21:43:17 |  Received VarSet event from AMI
20-Jun-2012 21:43:17 |    Value of event[variable] = RTPAUDIOQOSLOSS
20-Jun-2012 21:43:17 |    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;
20-Jun-2012 21:43:17 |  Received VarSet event from AMI
20-Jun-2012 21:43:17 |    Value of event[variable] = RTPAUDIOQOSRTT
20-Jun-2012 21:43:17 |    Value of event[value] = minrtt=0.000000;maxrtt=0.000000;avgrtt=0.000000;stdevrtt=0.000000;
20-Jun-2012 21:43:17 |  Received VarSet event from AMI
20-Jun-2012 21:43:17 |    Value of event[variable] = RTPAUDIOQOS
20-Jun-2012 21:43:17 |    Value of event[value] = ssrc=1268035130;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000
20-Jun-2012 21:43:17 |  self.connected_chan1:False
20-Jun-2012 21:43:17 |  self.connected_srtp1:False
20-Jun-2012 21:43:17 |  self.not_acceptable1:True
20-Jun-2012 21:43:17 |  self.connected_chan2:False
20-Jun-2012 21:43:17 |  self.connected_srtp2:False
20-Jun-2012 21:43:17 |  Test passed
20-Jun-2012 21:43:17 |  --> Running test 'tests/channels/SIP/secure_bridge_media' ...
20-Jun-2012 21:43:17 |  
20-Jun-2012 21:43:17 |  Making sure Asterisk isn't running ...

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