[asterisk-dev] [Code Review] Digium Phones "send to voicemail" tests

Matt Jordan reviewboard at asterisk.org
Thu May 17 12:15:00 CDT 2012


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/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uac/run-test
<https://reviewboard.asterisk.org/r/1926/#comment11560>

    Globally, don't use print - send it to the logger (probably as an INFO statement here).
    
    This would apply to all tests in this review.



/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uac/run-test
<https://reviewboard.asterisk.org/r/1926/#comment11572>

    You actually don't need the variable df here - you can just write:
    
    self.uas.run(self).addCallback(self.sippComplete)
    
    This would apply to your other tests as well.



/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uac/run-test
<https://reviewboard.asterisk.org/r/1926/#comment11561>

    This is already logged at a lower layer.



/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uac/test-config.yaml
<https://reviewboard.asterisk.org/r/1926/#comment11571>

    This needs some version tags of some sort.  This would apply to your other tests in here as well.


- Matt


On May 17, 2012, 10:10 a.m., Mark Michelson wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1926/
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> 
> (Updated May 17, 2012, 10:10 a.m.)
> 
> 
> Review request for Asterisk Developers, Jason Parker and Matt Jordan.
> 
> 
> Summary
> -------
> 
> First, see https://reviewboard.asterisk.org/r/1925 for an explanation of the "send to voicemail" Digium phones feature and how it is implemented.
> 
> This tests three scenarios.
> 
> 1) A called phone forwards call to voicemail.
> 2) On an established call, the calling party blind transfers the called party to voicemail.
> 3) On an established call, the called party blind transfers the calling party to voicemail.
> 
> In all three cases, the tests' passing is based on REDIRECTING(reason) evaluating to "send_to_vm" when the call is forwarded/transferred.
> 
> 
> Diffs
> -----
> 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uac/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uac/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uac/run-test PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uac/sipp/uac-blind-transfer.xml PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uac/sipp/uas.xml PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uac/test-config.yaml PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uas/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uas/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uas/run-test PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uas/sipp/uac-no-hangup.xml PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uas/sipp/uas-blind-transfer.xml PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uas/test-config.yaml PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/call_forward/configs/ast1/extensions.conf PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/call_forward/configs/ast1/sip.conf PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/call_forward/run-test PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/call_forward/sipp/uac.xml PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/call_forward/sipp/uas-redir.xml PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/call_forward/test-config.yaml PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/tests.yaml PRE-CREATION 
>   /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/tests.yaml 3225 
> 
> Diff: https://reviewboard.asterisk.org/r/1926/diff
> 
> 
> Testing
> -------
> 
> The tests all passed when combined with the Asterisk changes at https://reviewboard.asterisk.org/r/1925
> 
> 
> Thanks,
> 
> Mark
> 
>

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