> On Jan. 5, 2015, 10:47 p.m., Mark Michelson wrote:
> > branches/13/res/res_pjsip/include/res_pjsip_private.h, lines 123-141
> > <https://reviewboard.asterisk.org/r/4311/diff/1/?file=70133#file70133line123>
> >
> >     I unnastand the need for these functions, but I feel like they could be 
> > named more appropriately.
> >     
> >     First, due to a lack of proper namespacing in C, you have to be careful 
> > defining functions that are called pjsip_*. It is possible that such a 
> > function could be added to PJSIP itself and cause a conflict with us. Note 
> > that this applies to other functions besides the ones I've highlighted here.
> >     
> >     Second, the names do not do a great job of differentiating between the 
> > ast_sip_* versions of the functions. This may be hard to describe in a 
> > brief function name.
> 
> Kevin Harwell wrote:
>     How does sip_register_service_no_ref sound?  I'd like to avoid prefixing 
> it with "ast_" since that is usually reserved for exported functions and 
> these are internal.
>     
>     if sip_register_blah is too generic then maybe internal_sip_register?  
> Thoughts/suggestions?

The "internal" version sounds fine to me.


- Mark


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4311/#review14076
-----------------------------------------------------------


On Jan. 2, 2015, 7:46 p.m., Kevin Harwell wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4311/
> -----------------------------------------------------------
> 
> (Updated Jan. 2, 2015, 7:46 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-24485
>     https://issues.asterisk.org/jira/browse/ASTERISK-24485
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> The res_pjsip module was previously unloadable. With this patch it can now be 
> unloaded.
> 
> This patch is based off the original patch on the issue by Corey Farrell with 
> a few modifications. Removed a few changes not required to make the module 
> unloadable and also fixed a bug that would cause asterisk to crash on 
> unloading.
> 
> 
> Diffs
> -----
> 
>   branches/13/res/res_pjsip/pjsip_outbound_auth.c 430178 
>   branches/13/res/res_pjsip/pjsip_options.c 430178 
>   branches/13/res/res_pjsip/pjsip_global_headers.c 430178 
>   branches/13/res/res_pjsip/pjsip_distributor.c 430178 
>   branches/13/res/res_pjsip/pjsip_configuration.c 430178 
>   branches/13/res/res_pjsip/location.c 430178 
>   branches/13/res/res_pjsip/include/res_pjsip_private.h 430178 
>   branches/13/res/res_pjsip/config_transport.c 430178 
>   branches/13/res/res_pjsip/config_auth.c 430178 
>   branches/13/res/res_pjsip.c 430178 
>   branches/13/main/stasis_message_router.c 430178 
> 
> Diff: https://reviewboard.asterisk.org/r/4311/diff/
> 
> 
> Testing
> -------
> 
> Made it so res_pjsip was the only pjsip module loaded and then issued an 
> unload and noted it unloaded successfully (also loaded/unloaded it several 
> times from the CLI). Also when loaded and with REF_DEBUG enabled issued a 
> "core stop gracefully" and made sure there were no ref leaks for the module.
> 
> Also tested unloading with other dependent pjsip modules loaded and noted 
> that the module would not unload (as it should since dependencies are 
> currently loaded). And then shutdown asterisk and made sure it did not crash 
> or anything.
> 
> Started asterisk with nominal and off nominal module and pjsip configurations 
> to make sure things behaved appropriately (no crashes and such) and then 
> attempted to, or successfully unload the res_pjsip module. Also made sure 
> Asterisk continued to shutdown without incident.
> 
> 
> Thanks,
> 
> Kevin Harwell
> 
>

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to