[asterisk-dev] [Code Review] Add SIP Hold tests

[Paul Belanger]

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Ship it!


Really nice work!  I wish we added tests for all regressions :)


/asterisk/trunk/tests/channels/SIP/sip_hold/run-test
<https://reviewboard.asterisk.org/r/1647/#comment9352>

    add to sip.conf?



/asterisk/trunk/tests/channels/SIP/sip_hold/run-test
<https://reviewboard.asterisk.org/r/1647/#comment9350>

    why is this needed?



/asterisk/trunk/tests/channels/SIP/sip_hold/run-test
<https://reviewboard.asterisk.org/r/1647/#comment9351>

    can be removed, I believe create_ami_factory() already has a debug message.  If not, we should add it


- Paul


On Dec. 29, 2011, 10:19 a.m., Matt Jordan wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
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> 
> (Updated Dec. 29, 2011, 10:19 a.m.)
> 
> 
> Review request for Asterisk Developers, otherwiseguy, schmidts, and jrose.
> 
> 
> Summary
> -------
> 
> This adds a series of tests to the testsuite that cover SIP hold.  The tests use two phones (A / B), wherein Phone A calls Phone B, Phone B puts Phone A on hold, waits a period of time, removes the hold on Phone A, then sends a BYE.  The test checks that MOH is started / stopped in each scenario by subscribing to the MusicOnHold AMI event.
> 
> The SIP hold tests include Phone B sending a re-INVITE containing in the SDP either a restricted audio direction, a receiving IP address of 0.0.0.0, or a combination thereof.  The tests cover the two SIP endpoints interacting either directly or through a local RTP bridge.
> 
> Note that this test will fail in 1.8.8.0, and was used to test the regression identified in ASTERISK-19095.  The test will pass in 1.8.7.2, 1.8.8.1, and the current 1.8 branch.
> 
> 
> This addresses bug ASTERISK-19095.
>     https://issues.asterisk.org/jira/browse/ASTERISK-19095
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/channels/SIP/tests.yaml 2951 
>   /asterisk/trunk/tests/channels/SIP/sip_hold/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bridge.csv PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bypass.csv PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/lib/python/asterisk/sipp.py 2951 
> 
> Diff: https://reviewboard.asterisk.org/r/1647/diff
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> Matt
> 
>

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