[asterisk-dev] [Code Review]: Add mute all participants; play participant count to ConfBridge

[mjordan]

> On Nov. 3, 2011, 10:15 a.m., David Vossel wrote:
> > /trunk/apps/app_confbridge.c, lines 1576-1581
> > <https://reviewboard.asterisk.org/r/1518/diff/1/?file=21152#file21152line1576>
> >
> >     The channel isn't hearing the sound from the play_sound_helper() function because the channel is temporarily suspended from the conference when it enters an DTMF menu actions.  This is kind of weird, but we need to put something called autoservice on the channel here while we play back the sound file to the rest of the conference.
> >     
> >     Auto service will just make sure the read queue on the channel doesn't build up.  It basically just reads audio off the channel and discards it.

Will do.  I'll post a patch up once I've tested it with autoservice.


- mjordan


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On Oct. 11, 2011, 2:39 p.m., mjordan wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1518/
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> 
> (Updated Oct. 11, 2011, 2:39 p.m.)
> 
> 
> Review request for Asterisk Developers and David Vossel.
> 
> 
> Summary
> -------
> 
> This patch adds two new menu features to ConfBridge, admin_toggle_menu_participants and participant_count.  The admin action will globally mute / unmute all participants on a conference, while the participant count simply exposes the existing participant count function to the conference bridge menu.
> 
> Note that this is a port of the patch supplied by Kevin Reeves on ASTERISK-18204.  Very minor modifications were made to that patch, including adding sound file overriding to the config parser and some general overall cleanup.
> 
> This also makes a minor change that outputs the caller ID of the participants when the CLI command to list a specific conf bridge is used.  As this is useful and a minor modification, it was left in with this patch.
> 
> 
> This addresses bug ASTERISK-18204.
>     https://issues.asterisk.org/jira/browse/ASTERISK-18204
> 
> 
> Diffs
> -----
> 
>   /trunk/apps/confbridge/include/confbridge.h 340278 
>   /trunk/configs/confbridge.conf.sample 340278 
>   /trunk/apps/confbridge/conf_config_parser.c 340278 
>   /trunk/CHANGES 340278 
>   /trunk/apps/app_confbridge.c 340278 
> 
> Diff: https://reviewboard.asterisk.org/r/1518/diff
> 
> 
> Testing
> -------
> 
> Tested using a polycom phone and zoiper softphone.  No sound file quality issues were detecting in playing back the global sound using the conference bridge channel, which was an issue Kevin originally reported with his first implementation.  There did not appear to be any locking issues with the implementation proposed by Kevin.
> 
> 
> Thanks,
> 
> mjordan
> 
>

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