[asterisk-dev] [Code Review] 2592: testsuite: Create a basic test for call pickup

svnbot reviewboard at asterisk.org
Tue Jul 2 18:02:24 CDT 2013

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(Updated July 2, 2013, 6:02 p.m.)


This change has been marked as submitted.

Review request for Asterisk Developers, kmoore and Matt Jordan.


Committed in revision 3877

Bugs: ASTERISK-21544

Repository: testsuite


This test is based roughly on the directed pickup application test (but that test doesn't really work anymore). Basically it does the following:

1) A local channel is originated to an extension that will dial a SIP peer (faker). This SIP peer points to an unused address, so it won't answer.
2) Once the dial starts, Asterisk 2 dials Asterisk 1 via SIP to extension *8 (the pickup extension assigned in features.conf)
3) At this point the SIP channel on Asterisk 2 (sip_receive is its peername on Asterisk 1) should answer the call made by the local channel since faker is in a callgroup set for use by sip_receive. Both channels will then enter a simple bridge together.

Once https://reviewboard.asterisk.org/r/2582/ is committed I can also add the pickupsound being played as a condition for completing the test.


  /asterisk/trunk/tests/feature_call_pickup/configs/ast1/extensions.conf PRE-CREATION 
  /asterisk/trunk/tests/feature_call_pickup/configs/ast1/features.conf PRE-CREATION 
  /asterisk/trunk/tests/feature_call_pickup/configs/ast1/sip.conf PRE-CREATION 
  /asterisk/trunk/tests/feature_call_pickup/configs/ast2/extensions.conf PRE-CREATION 
  /asterisk/trunk/tests/feature_call_pickup/configs/ast2/sip.conf PRE-CREATION 
  /asterisk/trunk/tests/feature_call_pickup/run-test PRE-CREATION 
  /asterisk/trunk/tests/feature_call_pickup/test-config.yaml PRE-CREATION 
  /asterisk/trunk/tests/tests.yaml 3820 

Diff: https://reviewboard.asterisk.org/r/2592/diff/


Ran it a few times and made sure what was happening matched with log messages after the test was over. Made sure all pass conditions were actually met as well.



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