[asterisk-dev] [Code Review] 2383: Refactor Dialing to publish Stasis-Core messages

[David Lee]

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Not many issues; mostly minor.


/trunk/apps/app_dial.c
<https://reviewboard.asterisk.org/r/2383/#comment15750>

    This is a copy of the function in main/dial.c. Tsk, tsk.



/trunk/include/asterisk/channel.h
<https://reviewboard.asterisk.org/r/2383/#comment15747>

    Seriously, you've got an encoding problem here. Not sure what encoding your editor is using, but everything else seems to be expecting UTF-8.



/trunk/include/asterisk/stasis_channels.h
<https://reviewboard.asterisk.org/r/2383/#comment15749>

    In extracting this stuff to its own file, you dropped the '\addtogroup StasisTopicsAndMessages' doxygen tag I had added to trunk. Please put the back :-)



/trunk/main/dial.c
<https://reviewboard.asterisk.org/r/2383/#comment15751>

    Stray s at the end of the comment.



/trunk/main/manager_channels.c
<https://reviewboard.asterisk.org/r/2383/#comment15752>

    You shouldn't need both the suffix and the channel name. Just one should suffice.



/trunk/tests/test_stasis_channels.c
<https://reviewboard.asterisk.org/r/2383/#comment15753>

    \o/ Unit Tests!


- David Lee


On March 31, 2013, 12:29 p.m., Matt Jordan wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2383/
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> 
> (Updated March 31, 2013, 12:29 p.m.)
> 
> 
> Review request for Asterisk Developers and David Lee.
> 
> 
> Bugs: ASTERISK-21196
>     https://issues.asterisk.org/jira/browse/ASTERISK-21196
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> As part of the CDR work for Asterisk 12 (see https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification), we need Dial information published over Stasis-Core. This patch refactors app_dial to publish the necessary events.
> 
> The Dial based events are slightly different in Asterisk 12 than in other previous versions:
>  * Dial is now two events, DialBegin and DialEnd. This matches the nomenclature of other AMI events. See https://wiki.asterisk.org/wiki/display/AST/AMI+1.4+Specification for more information.
>  * Dial events now occur at the beginning of a dial operation and when the status of the dial operation is known. Previously, it occurred on application exit, which occurred after bridging.
> 
> Note that other applications will need refactoring as well (such as the Dial Framework, Queue, FollowMe, etc.) - however, in order to limit the scope of the work, I've kept it only to app_dial at this point.
> 
> 
> Diffs
> -----
> 
>   /trunk/main/manager_channels.c 384409 
>   /trunk/main/features.c 384409 
>   /trunk/main/dial.c 384409 
>   /trunk/main/channel_internal_api.c 384409 
>   /trunk/main/channel.c 384409 
>   /trunk/include/asterisk/stasis_channels.h PRE-CREATION 
>   /trunk/include/asterisk/channel.h 384409 
>   /trunk/apps/app_userevent.c 384409 
>   /trunk/apps/app_dial.c 384409 
>   /trunk/main/pbx.c 384409 
>   /trunk/main/stasis_channels.c PRE-CREATION 
>   /trunk/pbx/pbx_realtime.c 384409 
>   /trunk/tests/test_stasis_channels.c PRE-CREATION 
> 
> Diff: https://reviewboard.asterisk.org/r/2383/diff/
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> Matt Jordan
> 
>

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