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

[David Lee]

> On April 1, 2013, 2:07 p.m., David Lee wrote:
> > /trunk/main/manager_channels.c, lines 288-289
> > <https://reviewboard.asterisk.org/r/2383/diff/6/?file=35141#file35141line288>
> >
> >     You shouldn't need both the suffix and the channel name. Just one should suffice.
> 
> Matt Jordan wrote:
>     I have mixed feelings about this. Say we have two channels, one with a suffix of 'Dest':
>     
>     Channel: SIP/foo
>     ChannelDest: SIP/bar
>     Uniqueid: 1234
>     UniqueidDest: 5678
>     ChanVariable(SIP/foo): baz=shmackity
>     ChanVariable(SIP/bar): baz=yackity
>     
>     While the suffix is not strictly necessary, the lack of the suffix certainly makes the syntax less consistent. Are we sure this is preferable?

I'd be fine with going with just a suffix, since that's more consistent. It also avoids parsing problems when there's a ':' in the channel name.

Just be sure to document that in the CHANGES file, since that's yet another change to the AMI protocol in this release.


- David


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2383/#review8165
-----------------------------------------------------------


On April 1, 2013, 10:34 p.m., Matt Jordan wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2383/
> -----------------------------------------------------------
> 
> (Updated April 1, 2013, 10:34 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/tests/test_stasis_channels.c PRE-CREATION 
>   /trunk/pbx/pbx_realtime.c 384513 
>   /trunk/main/stasis_channels.c PRE-CREATION 
>   /trunk/main/pbx.c 384513 
>   /trunk/main/manager_channels.c 384513 
>   /trunk/main/features.c 384513 
>   /trunk/main/dial.c 384513 
>   /trunk/main/channel_internal_api.c 384513 
>   /trunk/main/channel.c 384513 
>   /trunk/include/asterisk/stasis_channels.h PRE-CREATION 
>   /trunk/include/asterisk/channel.h 384513 
>   /trunk/apps/app_userevent.c 384513 
>   /trunk/apps/app_dial.c 384513 
> 
> Diff: https://reviewboard.asterisk.org/r/2383/diff/
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> Matt Jordan
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130402/eaa74758/attachment.htm>