[asterisk-dev] [Code Review]: Digium Phones "send to voicemail" tests
Mark Michelson
reviewboard at asterisk.org
Thu May 17 15:48:44 CDT 2012
> On May 17, 2012, 12:15 p.m., Matt Jordan wrote:
> > /asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/redirecting_reason/blind_transfer_uac/test-config.yaml, lines 8-12
> > <https://reviewboard.asterisk.org/r/1926/diff/2/?file=28042#file28042line8>
> >
> > This needs some version tags of some sort. This would apply to your other tests in here as well.
Any guidance on what sort of version tags to add? What is the current method of indicating that I want this to run for all certified Asterisk branches from 1.8.11 forward, and for 10-digiumphones, and for Asterisk 11, but not for vanilla 1.8 or vanilla 10?
- Mark
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1926/#review6235
-----------------------------------------------------------
On May 17, 2012, 10:10 a.m., Mark Michelson wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1926/
> -----------------------------------------------------------
>
> (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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20120517/0fe9a410/attachment.htm>
More information about the asterisk-dev
mailing list