[asterisk-dev] [Code Review] 3970: res_phoneprov: Refactor phoneprov to allow pluggable config providers.

[George Joseph]

> On Sept. 18, 2014, 4:11 p.m., rmudgett wrote:
> > branches/12/res/res_phoneprov.c, lines 128-129
> > <https://reviewboard.asterisk.org/r/3970/diff/4/?file=67281#file67281line128>
> >
> >     This is an interesting way of doing the standard ao2 container callback functions.
> >     
> >     Did you fix the formatting from the wiki page.  Directly cutting and pasting the template function from the wiki gives you spaces instead of tabs.

Yep, I reformatted it after pasting it.


> On Sept. 18, 2014, 4:11 p.m., rmudgett wrote:
> > branches/12/res/res_phoneprov.c, line 983
> > <https://reviewboard.asterisk.org/r/3970/diff/4/?file=67281#file67281line983>
> >
> >     Is the cast necessary?

Yes.  The compiler complains 
note: expected ‘void *’ but argument is of type ‘ast_string_field’


- George


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On Sept. 18, 2014, 6:01 p.m., George Joseph wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3970/
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> 
> (Updated Sept. 18, 2014, 6:01 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> The big piece missing for me to finally transition to pjsip was the ability to mirror the auto provisioning features of res_phoneprov.  The first step (this patch) is to make res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf.  To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions.
> 
> ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users.
> ast_phoneprov_provider_unregister clears the defaults and users.
> ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them.
> ast_phoneprov_delete_extension deletes one extension.
> ast_phoneprov_delete_extensions deletes all extensions for the provider.
> 
> res_phoneprov actually registers itself as the provider for sip/users and is always available and is the default.
> 
> Writing a new provider...
> Since res_phoneprov is also it's own provider, examples of what a new provider would have to do are in load_users() in res_phoneprov.c.  Those functions gather the information from users.conf and sip.conf and call the ast_provider_register and ast_phoneprov_add_extension apis.
> 
> So...
> The provider creates a callback function which calls the ast_phoneprov_add_extension api for each user.  
> It then calls ast_phoneprov_provider_register with the callback.
> res_phoneprov then calls the callback to cause the actual load.
> During normal http server ops, all work is done by res_phoneprov and the provider is never called again unless a reload is needed.
> If the provider wants to reload it can simply unregister and reregister or it can call its own load_users callback.
> If res_phoneprov wants to reload, it iterates over its registry and calls the providers callback.
> 
> NOTE:  If res_phoneprov is actually unloaded, it has no way to know what providers were registered (other than itself) so a subsequent load will have nothing but it's own users.  
> 
> Additional changes...
> I added a few convenience functions to chanvars for creating lists and finding and deleting entries.  No existing code was touched.
> 
> Next steps...
> A provider for res_pjsip.
> 
> 
> Diffs
> -----
> 
>   branches/12/res/res_phoneprov.exports.in PRE-CREATION 
>   branches/12/res/res_phoneprov.c 423501 
>   branches/12/main/chanvars.c 423501 
>   branches/12/include/asterisk/phoneprov.h PRE-CREATION 
>   branches/12/include/asterisk/chanvars.h 423501 
>   branches/12/configs/phoneprov.conf.sample 423501 
> 
> Diff: https://reviewboard.asterisk.org/r/3970/diff/
> 
> 
> Testing
> -------
> 
> I ran through several scenarios including the use of PP_EACH_USER and PP_EACH_EXTENSION to make sure that all existing functionality was preserved.  I actually use it with Grandstream phones and everything worked exactly as expected.
> 
> 
> Thanks,
> 
> George Joseph
> 
>

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