[asterisk-dev] [Code Review] SIP Blind transfer tests

Mark Michelson reviewboard at asterisk.org
Mon Jan 23 16:44:36 CST 2012


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Ship it!


Looking good to me.


/asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_refer_only/run-test
<https://reviewboard.asterisk.org/r/1686/#comment9827>

    Since the minversion of this test is 1.8.9.0, there's no need to do this link checking here. That's something that has not existed since Asterisk 1.4. This applies to all tests.


- Mark


On Jan. 23, 2012, 3:46 p.m., Matt Jordan wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1686/
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> 
> (Updated Jan. 23, 2012, 3:46 p.m.)
> 
> 
> Review request for Asterisk Developers and Mark Michelson.
> 
> 
> Summary
> -------
> 
> This patch adds four SIP blind transfer tests to the testsuite.  For Phone A, Phone B, and Phone C, where Phone A initially calls Phone B, they test:
> 1. Phone A initiating a blind transfer of Phone B to Phone C with no re-INVITE prior to the REFER message
> 2. Phone A initiating a blind transfer of Phone B to Phone C with a re-INVITE
> 3. Phone B initiating a blind transfer of Phone A to Phone C with no re-INVITE prior to the REFER message
> 4. Phone B initiating a blind transfer of Phone A to Phone C with a re-INVITE
> 
> Note that adding the 'h' extension currently reproduces the bug (ASTERISK-19173) fixed by Mark on patch https://reviewboard.asterisk.org/r/1685/, hence its inclusion in the test.
> 
> 
> This addresses bug ASTERISK-19173.
>     https://issues.asterisk.org/jira/browse/ASTERISK-19173
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_refer_only/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_refer_only/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_refer_only/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_refer_only/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_with_reinvite/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_refer_only/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_refer_only/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_refer_only/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_refer_only/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/caller_with_reinvite/test-config.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/sip_blind_transfer/tests.yaml PRE-CREATION 
>   /asterisk/trunk/tests/channels/SIP/tests.yaml 3000 
> 
> Diff: https://reviewboard.asterisk.org/r/1686/diff
> 
> 
> Testing
> -------
> 
> Need to test with Mark's patch - without the patch, Scenarios 3 and 4 will fail, while Scenario 1 results in an orphaned bridge.  It is expected that the patch resolves all three of those issues.
> 
> 
> Thanks,
> 
> Matt
> 
>

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