[asterisk-dev] mmichelson: trunk r157427 - /trunk/channels/chan_sip.c
russell at digium.com
Tue Nov 18 14:37:53 CST 2008
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
> * 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
> hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586)
> 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 */
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
Senior Software Engineer
Open Source Team Lead
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