[asterisk-dev] [Code Review] 2627: Refactor extraneous channel events
Matt Jordan
reviewboard at asterisk.org
Mon Jun 24 19:58:44 CDT 2013
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trunk/channels/chan_dahdi.c
<https://reviewboard.asterisk.org/r/2627/#comment17629>
Same finding about DAHDI channels versus Asterisk channels.
trunk/channels/chan_dahdi.c
<https://reviewboard.asterisk.org/r/2627/#comment17627>
Since we have the capability to do it, I'd add a <note> element here that explicitly says this is not an Asterisk channel.
"Channel" as an AMI event key has a particularly important connotation, and if we're going to repurpose it we should be rather obvious that we're doing so.
(Alternatively, rename this to DAHDIChannel)
trunk/channels/chan_dahdi.c
<https://reviewboard.asterisk.org/r/2627/#comment17628>
Same note here. Also call out that this is a DAHDI channel.
trunk/channels/chan_dahdi.c
<https://reviewboard.asterisk.org/r/2627/#comment17630>
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trunk/channels/chan_dahdi.c
<https://reviewboard.asterisk.org/r/2627/#comment17631>
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trunk/channels/chan_dahdi.c
<https://reviewboard.asterisk.org/r/2627/#comment17632>
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trunk/channels/chan_dahdi.c
<https://reviewboard.asterisk.org/r/2627/#comment17633>
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trunk/channels/chan_dahdi.c
<https://reviewboard.asterisk.org/r/2627/#comment17626>
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trunk/channels/chan_gtalk.c
<https://reviewboard.asterisk.org/r/2627/#comment17635>
So, I think I forgot to include chan_iax or chan_sip's ChannelUpdate events in this issue. Whoops.
On the other hand, the more I look at the ChannelUpdate events, the more I wonder what purpose they have. You can't use the GTalk SID for anything - there aren't any AMI actions that consume it.
In chan_sip, we output the fullcontact as well as the callid. I'm not sure what you'd do with that externally either.
chan_iax is the same (callids).
The useful thing chan_iax emits, however, is an association between the channel and the IAX peer. I think we could get that, however, from the creation of channels and their association with Stasis endpoints.
Do we really need these events?
trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/2627/#comment17636>
Document this as an enum, with possible values RTPTimeout and SIPSessionTimer
trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/2627/#comment17634>
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trunk/channels/sig_analog.c
<https://reviewboard.asterisk.org/r/2627/#comment17637>
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trunk/channels/sig_analog.c
<https://reviewboard.asterisk.org/r/2627/#comment17638>
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- Matt Jordan
On June 19, 2013, 8:25 p.m., opticron wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2627/
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>
> (Updated June 19, 2013, 8:25 p.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Bugs: ASTERISK-21476
> https://issues.asterisk.org/jira/browse/ASTERISK-21476
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> This change removes JitterBufStats and ChannelReload and refactors the following events to travel over Stasis-Core:
> * LocalBridge
> * DAHDIChannel
> * AlarmClear
> * SpanAlarmClear
> * Alarm
> * SpanAlarm
> * DNDState
> * MCID
> * ChannelUpdate
> * SIPQualifyPeerDone
> * SessionTimeout
>
>
> Diffs
> -----
>
> trunk/CHANGES 392240
> trunk/apps/app_meetme.c 392240
> trunk/apps/app_queue.c 392240
> trunk/channels/chan_dahdi.c 392240
> trunk/channels/chan_gtalk.c 392240
> trunk/channels/chan_iax2.c 392240
> trunk/channels/chan_sip.c 392240
> trunk/channels/sig_analog.c 392240
> trunk/channels/sig_pri.c 392240
> trunk/include/asterisk/json.h 392240
> trunk/include/asterisk/stasis.h 392240
> trunk/main/core_local.c 392240
> trunk/main/json.c 392240
> trunk/res/res_agi.c 392240
>
> Diff: https://reviewboard.asterisk.org/r/2627/diff/
>
>
> Testing
> -------
>
> Manual testing. Automated tests coming for some.
>
>
> Thanks,
>
> opticron
>
>
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