> On Feb. 10, 2015, 12:29 p.m., Mark Michelson wrote: > > 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 :)
If this turns out to be a problem then we can change the Wait() back to 5 seconds. If that is the case then I think it will be appropriate to increase the timeout for 'core stop gracefully' from 5 seconds to 10. - Corey ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4411/#review14433 ----------------------------------------------------------- On Feb. 9, 2015, 12:50 p.m., Corey Farrell wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4411/ > ----------------------------------------------------------- > > (Updated Feb. 9, 2015, 12:50 p.m.) > > > Review request for Asterisk Developers. > > > Repository: testsuite > > > Description > ------- > > * 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. > > > Diffs > ----- > > /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 > > Diff: https://reviewboard.asterisk.org/r/4411/diff/ > > > Testing > ------- > > 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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