[asterisk-dev] [Code Review] 4411: testsuite: fix a number of tests where Asterisk does not shutdown gracefully

[Mark Michelson]

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Ship it!


The only caveat here is that you may want to watch automated runs of the SIP info_dtmf test to be sure that on awful hardware the 5 second Wait() isn't too short for the test to complete. I suspect it will be okay though.

On a side note, I have another test to add to my list of tests that could be rewritten to not rely on timing, though :)

- Mark Michelson


On Feb. 9, 2015, 5:50 p.m., Corey Farrell wrote:
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> https://reviewboard.asterisk.org/r/4411/
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> (Updated Feb. 9, 2015, 5:50 p.m.)
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> 
> Review request for Asterisk Developers.
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> 
> Repository: testsuite
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> 
> Description
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> * Add Hangup() to priority after Dial() where needed.  This prevents auto-fallthrough from playing 10 seconds of BUSY or CONGESTION tone.
> * Decrease Wait(10) to Wait(5) in tests/channels/SIP/info_dtmf.
> * Maintain list of AGI connections where needed so they can all be agi.finish().
> * Replace calls to reactor.stop() with self.stop_reactor(), remove test.start_asterisk()/test.stop_asterisk() from main().
> * Delay self.stop_reactor() in tests/channels/SIP/sip_tls_call by 2 seconds.  This gives the calls enough time to end and avoid shutdown timeout.
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> 
> Diffs
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> 
>   /asterisk/trunk/tests/funcs/func_srv/run-test 6377 
>   /asterisk/trunk/tests/funcs/func_presencestate/run-test 6377 
>   /asterisk/trunk/tests/fastagi/stream-file/run-test 6377 
>   /asterisk/trunk/tests/fastagi/database/run-test 6377 
>   /asterisk/trunk/tests/fastagi/control-stream-file/run-test 6377 
>   /asterisk/trunk/tests/channels/SIP/sip_tls_call/run-test 6377 
>   /asterisk/trunk/tests/channels/SIP/sip_srtp/run-test 6377 
>   /asterisk/trunk/tests/channels/SIP/sip_cause/configs/ast1/extensions.conf 6377 
>   /asterisk/trunk/tests/channels/SIP/secure_bridge_media/run-test 6377 
>   /asterisk/trunk/tests/channels/SIP/noload_res_srtp/run-test 6377 
>   /asterisk/trunk/tests/channels/SIP/info_dtmf/configs/ast1/extensions.conf 6377 
>   /asterisk/trunk/tests/channels/SIP/hangupcause/configs/ast1/extensions.conf 6377 
>   /asterisk/trunk/tests/channels/SIP/generic_ccss/configs/ast1/extensions.conf 6377 
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> Diff: https://reviewboard.asterisk.org/r/4411/diff/
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> 
> Testing
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> Ran all effected tests against Asterisk 11 with REF_DEBUG.  Prior to these fixes graceful shutdown of Asterisk timed out, causing reference leaks to be reported.  These tests now shutdown gracefully and have no reference leaks.
> 
> 
> Thanks,
> 
> Corey Farrell
> 
>

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