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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4411/
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Review request for Asterisk Developers.


Repository: testsuite


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.


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 

Diff: https://reviewboard.asterisk.org/r/4411/diff/


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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