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

[George Joseph]

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https://reviewboard.asterisk.org/r/3970/
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(Updated Sept. 5, 2014, 6:01 p.m.)


Review request for Asterisk Developers.


Changes
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Removed the load_defaults process.  It's actually easier to let the provider load and apply it's own defaults.
Cleaned up some error messages.
Added language to phoneprov.conf.sample concerning defaults and providers.


Repository: Asterisk


Description (updated)
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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.

Since res_phoneprov is a module that may or may not be loaded, the apis are implemented as inlines that check that res_phoneprov is actually loaded before calling the concrete function.  A new include file was added: include/asterisk/phoneprov.h

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_defaults() and load_users() in res_phoneprov.c.  Those functions gather the information from users.conf and sip.conf and call the ast_provider_set_defaults 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 (updated)
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  branches/12/res/res_phoneprov.exports.in PRE-CREATION 
  branches/12/res/res_phoneprov.c 422737 
  branches/12/main/chanvars.c 422737 
  branches/12/include/asterisk/phoneprov.h PRE-CREATION 
  branches/12/include/asterisk/chanvars.h 422737 
  branches/12/configs/phoneprov.conf.sample 422737 

Diff: https://reviewboard.asterisk.org/r/3970/diff/


Testing
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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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