[asterisk-dev] [Code Review] 3970: res_phoneprov: Refactor phoneprov to allow pluggable config providers.
opticron
reviewboard at asterisk.org
Tue Sep 30 10:20:53 CDT 2014
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3970/#review13404
-----------------------------------------------------------
branches/12/include/asterisk/phoneprov.h
<https://reviewboard.asterisk.org/r/3970/#comment23888>
Document the return value for this typedef.
branches/12/include/asterisk/phoneprov.h
<https://reviewboard.asterisk.org/r/3970/#comment23889>
This documentation conflicts with the prototype.
branches/12/main/chanvars.c
<https://reviewboard.asterisk.org/r/3970/#comment23890>
Note that there should be an empty line after declarations. Apply this below as well.
branches/12/res/res_phoneprov.c
<https://reviewboard.asterisk.org/r/3970/#comment23891>
s/whi/who/
branches/12/res/res_phoneprov.exports.in
<https://reviewboard.asterisk.org/r/3970/#comment23893>
This is unnecessary since there are no symbols that match it.
- opticron
On Sept. 18, 2014, 7:01 p.m., George Joseph wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3970/
> -----------------------------------------------------------
>
> (Updated Sept. 18, 2014, 7:01 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140930/89693a17/attachment-0001.html>
More information about the asterisk-dev
mailing list