[asterisk-dev] [Code Review] Add mutestream manager action and MUTESTREAM() dialplan function

[Russell Bryant]

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/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2489>

    Copyright (C) 2009, Olle E. Johansson



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2490>

    You probably don't need all of these included explicitly.



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2491>

    doxygen format :-)



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2492>

    Please use ast_free()



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2493>

    Take a look at the places where reviewboard has highlighted whitespace errors in red.



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2494>

    Use ast_debug()



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2495>

    It looks like you have an extra level of indentation here.



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2497>

    add a space after if



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2496>

    ast_debug()



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2498>

    You must lock the channel around datastore operations.  That applies here and some other places in this module.



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2499>

    The documentation will need to be in XML format.



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2500>

    There is a memory leak here.  The reference to the channel needs to be released.



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2501>

    AST_MODULE_LOAD_SUCCESS



/trunk/res/res_mutestream.c
<https://reviewboard.asterisk.org/r/345/#comment2502>

    Why not?


- Russell


On 2009-08-31 12:47:29, Olle E Johansson wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/345/
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> 
> (Updated 2009-08-31 12:47:29)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> When you need to be able to mute incoming or outgoing audio for a channel, this is the function you need.
> 
> I am a bit unsure of the unload functionality. What will happen if I unload an audiohook module when a channel is active? 
> 
> This work is inspired by app_jack.c and func_volume.c
> 
> 
> Diffs
> -----
> 
>   /trunk/res/res_mutestream.c PRE-CREATION 
> 
> Diff: https://reviewboard.asterisk.org/r/345/diff
> 
> 
> Testing
> -------
> 
> I've tested this in manager and with the dynamic features in features.conf.
> 
> 
> Thanks,
> 
> Olle E
> 
>