[asterisk-dev] mmichelson: trunk r157427 - /trunk/channels/chan_sip.c

[Russell Bryant]

SVN commits to the Digium repositories wrote:
> Author: mmichelson
> Date: Tue Nov 18 14:23:58 2008
> New Revision: 157427
> 
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=157427
> Log:
> * Add a lock to be used in the update_call_counter function.
> * Revert logic to mirror 1.4's in the sense that it will not allow
>   the call counter to dip below 0.
> 
> These two measures prevent potential races that could cause a SIP peer
> to appear to be busy forever.
> 
> (closes issue #13668)
> Reported by: mjc
> Patches:
>       hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586)
> 
> 
> Modified:
>     trunk/channels/chan_sip.c
> 
> Modified: trunk/channels/chan_sip.c
> URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=157427&r1=157426&r2=157427
> ==============================================================================
> --- trunk/channels/chan_sip.c (original)
> +++ trunk/channels/chan_sip.c Tue Nov 18 14:23:58 2008
> @@ -2118,6 +2118,9 @@
>  	SDP_T38_ACCEPT,   /*!< Remote side accepted our T38 request */
>  };
>  
> +/*! \brief Protect the callcounters inuse,inringing and the corresponding flags */
> +AST_MUTEX_DEFINE_STATIC(callctrlock);
> +

What is the logic behind using this new global lock for this purpose? 
What is wrong with just ensuring that the sip_pvt is locked during these 
operations?

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.