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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3970/">https://reviewboard.asterisk.org/r/3970/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By George Joseph.</div>
<p style="color: grey;"><i>Updated Sept. 5, 2014, 6:01 p.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Changes</h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description (updated)</h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> (updated)</h1>
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<li>branches/12/res/res_phoneprov.exports.in <span style="color: grey">(PRE-CREATION)</span></li>
<li>branches/12/res/res_phoneprov.c <span style="color: grey">(422737)</span></li>
<li>branches/12/main/chanvars.c <span style="color: grey">(422737)</span></li>
<li>branches/12/include/asterisk/phoneprov.h <span style="color: grey">(PRE-CREATION)</span></li>
<li>branches/12/include/asterisk/chanvars.h <span style="color: grey">(422737)</span></li>
<li>branches/12/configs/phoneprov.conf.sample <span style="color: grey">(422737)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3970/diff/" style="margin-left: 3em;">View Diff</a></p>
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