> 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 ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3970/#review13348 ----------------------------------------------------------- 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/ > ----------------------------------------------------------- > > (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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