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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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