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

[jrose]

> On June 24, 2013, 9:10 p.m., Matt Jordan wrote:
> > One of the primary reasons for having tests is to verify functionality that shouldn't change between versions. By having a test that runs against both 11 and 12, we can ensure that regardless of AMI event differences, configuration changes, etc. - the basic functionality still works. I really think this test needs to run against 11 and 12, which will probably change how you structure this test, since the AMI events have changed so significantly.
> > 
> > {quote}
> > This test is based roughly on the directed pickup application test (but that test doesn't really work anymore).
> > {quote}
> > 
> > That's a sign that something changed. Either that test is broken, or Pickup has changed enough that valid configurations will now break.

Pickup as a feature is distinct from app_pickup, which is what this was testing. It doesn't really have anything to do with the work I was doing for the Pickup feature. I've already fixed that test though, the problem was entirely with AMI events... specifically the difference between Bridge and BridgeEnter.


- jrose


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2592/#review8952
-----------------------------------------------------------


On June 5, 2013, 4:02 p.m., jrose wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2592/
> -----------------------------------------------------------
> 
> (Updated June 5, 2013, 4:02 p.m.)
> 
> 
> Review request for Asterisk Developers, kmoore and Matt Jordan.
> 
> 
> Bugs: ASTERISK-21544
>     https://issues.asterisk.org/jira/browse/ASTERISK-21544
> 
> 
> Repository: testsuite
> 
> 
> Description
> -------
> 
> 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.
> 
> 
> Diffs
> -----
> 
>   /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 
> 
> Diff: https://reviewboard.asterisk.org/r/2592/diff/
> 
> 
> Testing
> -------
> 
> 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.
> 
> 
> Thanks,
> 
> jrose
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130624/77d50534/attachment.htm>