[test-results] [Bamboo] Asterisk Testing - Asterisk Trunk - Asterisk CentOS 6 64-Bit 121 may have hung.

[Bamboo]

-------------------------------------------------------------------------------
TESTING-ASTERISKTRUNK-AST18CENTOS64-121 may have hung.
-------------------------------------------------------------------------------

This build has been running for 173 minutes, which is 149% longer than usual.
It has been 157 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

--------------
Code Changes
--------------
mmichelson (360886):

>Fix potential race condition during call pickup.
>
>Prior to this patch, a connected line update was queued during
>call pickup and then an answer frame was queued. The original
>caller would presumably then have his connected line updated
>and then the call would be answered.
>
>In actuality, the answer frame was not how the call ended up
>being answered. Rather, an odd section in app_dial that checks
>if the called channel's state is up.
>
>The result is that the order of the connected line update and
>the answer were variable. In most cases, this wasn't actually
>a bad thing. However, if the 'I' option was passed to dial, the
>connected line update would be inhibited.
>
>The fix is to queued the connected line after the answer frame is
>queued. This way the race in app_dial is between two
>conditions resulting in an answer. This way the connected line
>update occurs after the answer every time.
>
>(closes issue ASTERISK-19183)
>Reported by: Thomas Arimont
>Tested by: Thomas Arimont
>    Mark Michelson
>Patches:
>    ASTERISK-19183.patch uploaded by Mark Michelson (license 5049)
>........
>
>Merged revisions 360884 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 360885 from http://svn.asterisk.org/svn/asterisk/branches/10
>



--------------
Last Logs
--------------
30-Mar-2012 06:49:03 |  --> Cannot run test 'tests/channels/SIP/refer_replaces_to_self'
30-Mar-2012 06:49:03 |  --- --> Minimum Version: 1.4 (True)
30-Mar-2012 06:49:03 |  --- --> Tags: ['SIP', 'transfer']
30-Mar-2012 06:49:03 |  --- --> Dependency: pjsua - False
30-Mar-2012 06:49:03 |  
30-Mar-2012 06:49:03 |  --> Running test 'tests/channels/SIP/info_dtmf' ...
30-Mar-2012 06:49:03 |  
30-Mar-2012 06:49:03 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:49:03 |  Running ['tests/channels/SIP/info_dtmf/run-test'] ...
30-Mar-2012 06:49:03 |  --> Running test 'tests/channels/SIP/tcpauthlimit' ...
30-Mar-2012 06:49:03 |  
30-Mar-2012 06:49:03 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:49:03 |  Running ['tests/channels/SIP/tcpauthlimit/run-test'] ...
30-Mar-2012 06:49:03 |  starting asterisk
30-Mar-2012 06:49:03 |  connecting 5 clients to asterisk
30-Mar-2012 06:49:03 |  attempting to connect one more, this should fail
30-Mar-2012 06:49:03 |  connecting and authenticating 10 clients to asterisk
30-Mar-2012 06:49:03 |  checking for errors
30-Mar-2012 06:49:03 |  test passed
30-Mar-2012 06:49:03 |  --> Running test 'tests/channels/SIP/tcpauthtimeout' ...
30-Mar-2012 06:49:03 |  
30-Mar-2012 06:49:03 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:49:03 |  Running ['tests/channels/SIP/tcpauthtimeout/run-test'] ...
30-Mar-2012 06:49:03 |  starting asterisk
30-Mar-2012 06:49:03 |  testing timeout of an unauthenticated session
30-Mar-2012 06:49:03 |  testing timeout of an unauthenticated session after writing some data
30-Mar-2012 06:49:03 |  testing timeout of an unauthenticated session after writing some different data
30-Mar-2012 06:49:03 |  testing timeout of an unauthenticated session after writing data in bursts
30-Mar-2012 06:50:33 |  testing timeout of an authenticated session (should not timeout)
30-Mar-2012 06:50:33 |  test passed
30-Mar-2012 06:50:33 |  --> tests/channels/SIP/sip_outbound_address ... skipped 'Skip while failures are debugged'
30-Mar-2012 06:50:33 |  --> tests/channels/SIP/sip_attended_transfer ... skipped 'Skip while failures are debugged'
30-Mar-2012 06:50:33 |  --> tests/channels/SIP/sip_attended_transfer_tcp ... skipped 'Skip while failures are debugged'
30-Mar-2012 06:50:33 |  --> tests/channels/SIP/sip_attended_transfer_v6 ... skipped 'Skip while failures are debugged'
30-Mar-2012 06:50:33 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/callee_refer_only' ...
30-Mar-2012 06:50:33 |  
30-Mar-2012 06:50:33 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:50:33 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_refer_only/run-test'] ...
30-Mar-2012 06:50:33 |  [Mar 30 06:49:07] WARNING[23957]: __main__:79 bridge_event_handler: Unexpected bridgetype core or bridgestate unlink received!
30-Mar-2012 06:50:33 |  [Mar 30 06:49:07] WARNING[23957]: __main__:79 bridge_event_handler: Unexpected bridgetype core or bridgestate unlink received!
30-Mar-2012 06:50:33 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/callee_with_reinvite' ...
30-Mar-2012 06:50:33 |  
30-Mar-2012 06:50:33 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:50:33 |  Running ['tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/run-test'] ...
30-Mar-2012 06:50:33 |  [Mar 30 06:49:12] WARNING[24032]: __main__:79 bridge_event_handler: Unexpected bridgetype core or bridgestate unlink received!
30-Mar-2012 06:50:33 |  [Mar 30 06:49:12] WARNING[24032]: __main__:79 bridge_event_handler: Unexpected bridgetype core or bridgestate unlink received!
30-Mar-2012 06:50:33 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_refer_only' ...
30-Mar-2012 06:50:33 |  
30-Mar-2012 06:50:33 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:50:33 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_refer_only/run-test'] ...
30-Mar-2012 06:50:33 |  [Mar 30 06:49:18] WARNING[24106]: __main__:80 bridge_event_handler: Unexpected bridgetype core or bridgestate unlink received!
30-Mar-2012 06:50:33 |  [Mar 30 06:49:18] WARNING[24106]: __main__:80 bridge_event_handler: Unexpected bridgetype core or bridgestate unlink received!
30-Mar-2012 06:50:33 |  --> Running test 'tests/channels/SIP/sip_blind_transfer/caller_with_reinvite' ...
30-Mar-2012 06:50:33 |  
30-Mar-2012 06:50:33 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:50:33 |  Running ['tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/run-test'] ...
30-Mar-2012 06:50:33 |  [Mar 30 06:49:23] WARNING[24181]: __main__:86 bridge_event_handler: Unexpected bridgetype core or bridgestate unlink received!
30-Mar-2012 06:50:33 |  [Mar 30 06:49:23] WARNING[24181]: __main__:86 bridge_event_handler: Unexpected bridgetype core or bridgestate unlink received!
30-Mar-2012 06:50:33 |  --> Running test 'tests/channels/SIP/sip_one_legged_transfer' ...
30-Mar-2012 06:50:33 |  
30-Mar-2012 06:50:33 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:50:33 |  Running ['tests/channels/SIP/sip_one_legged_transfer/run-test'] ...
30-Mar-2012 06:50:33 |  [Mar 30 06:49:48] WARNING[24256]: __main__:64 checkBridgeResult: 'link' and 'bridgedchannel' not found
30-Mar-2012 06:50:33 |  --> tests/channels/SIP/sip_one_legged_transfer_v6 ... skipped 'Skip while failures are debugged'
30-Mar-2012 06:50:33 |  --> Running test 'tests/channels/SIP/sip_register' ...
30-Mar-2012 06:50:33 |  
30-Mar-2012 06:50:33 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:50:33 |  Running ['tests/channels/SIP/sip_register/run-test'] ...
30-Mar-2012 06:50:33 |  --> Running test 'tests/channels/SIP/sip_register_domain_acl' ...
30-Mar-2012 06:50:33 |  
30-Mar-2012 06:50:33 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:50:33 |  Running ['tests/channels/SIP/sip_register_domain_acl/run-test'] ...
30-Mar-2012 06:50:33 |  --> Running test 'tests/channels/SIP/sip_channel_params' ...
30-Mar-2012 06:50:33 |  
30-Mar-2012 06:50:33 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:50:33 |  Running ['tests/channels/SIP/sip_channel_params/run-test'] ...
30-Mar-2012 06:50:33 |  Got channel name SIP/test1-00000000
30-Mar-2012 06:50:33 |  test passed
30-Mar-2012 06:50:33 |  --> Running test 'tests/channels/SIP/sip_tls_call' ...
30-Mar-2012 06:50:33 |  
30-Mar-2012 06:50:33 |  Making sure Asterisk isn't running ...
30-Mar-2012 06:50:33 |  Running ['tests/channels/SIP/sip_tls_call/run-test'] ...
30-Mar-2012 06:50:33 |  Building test resources ...
30-Mar-2012 06:50:33 |  Creating Asterisk instances ...
30-Mar-2012 06:50:33 |  AMI 1 - connected, registering DTMF event...
30-Mar-2012 06:50:33 |  AMI - originating call from SIP/testast1 to extension 1000 using extension 1000 at priority 1
30-Mar-2012 06:50:33 |  AMI 2 - connected, registering DTMF event...
30-Mar-2012 06:50:33 |  Received DTMF event from AMI 1...
30-Mar-2012 06:50:33 |  Value of DTMF[digit] = 5
30-Mar-2012 06:50:33 |  It's a match for ast[0] receiving DTMF from ast[1]
30-Mar-2012 06:50:33 |  Received DTMF event from AMI 1...
30-Mar-2012 06:50:33 |  Value of DTMF[digit] = 5
30-Mar-2012 06:50:33 |  It's a match for ast[0] receiving DTMF from ast[1]
30-Mar-2012 06:50:33 |  Received DTMF event from AMI 2...
30-Mar-2012 06:50:33 |  Value of DTMF[digit] = 6
30-Mar-2012 06:50:33 |  It's a match for ast[1] receiving DTMF from ast[0]
30-Mar-2012 06:50:33 |  Both tones have been matched at least once.  Test PASSED.
30-Mar-2012 06:50:33 |  --> Running test 'tests/channels/SIP/sip_tls_register' ...
30-Mar-2012 06:50:33 |  
30-Mar-2012 06:50:33 |  Making sure Asterisk isn't running ...

--
This message is automatically generated by Atlassian Bamboo
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/test-results/attachments/20120330/698cabd3/attachment-0001.htm>