[asterisk-commits] rizzo: branch rizzo/astobj2 r47838 - /team/rizzo/astobj2/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: rizzo
Date: Sun Nov 19 03:56:12 2006
New Revision: 47838

URL: http://svn.digium.com/view/asterisk?view=rev&rev=47838
Log:
move sip_destroy() next to __sip_destroy() as they are very related.

I recommend this for trunk as well.


Modified:
    team/rizzo/astobj2/channels/chan_sip.c

Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=47838&r1=47837&r2=47838
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Sun Nov 19 03:56:12 2006
@@ -3218,6 +3218,16 @@
 	return NULL;
 }
 
+/*! \brief Destroy SIP call structure (after unlinking it from the list) */
+static struct sip_pvt *sip_destroy(struct sip_pvt *p)
+{
+	if (option_debug > 2)
+		ast_log(LOG_DEBUG, "Destroying SIP dialog %s\n", p->callid);
+	sip_pvt_unlink(p);
+	__sip_destroy(p);
+	return NULL;
+}
+
 /*! \brief  update_call_counter: Handle call_limit for SIP users 
  * Setting a call-limit will cause calls above the limit not to be accepted.
  *
@@ -3336,16 +3346,6 @@
 	} else /* u must be set */
 		unref_user(u);
 	return 0;
-}
-
-/*! \brief Destroy SIP call structure (after unlinking it from the list) */
-static struct sip_pvt *sip_destroy(struct sip_pvt *p)
-{
-	if (option_debug > 2)
-		ast_log(LOG_DEBUG, "Destroying SIP dialog %s\n", p->callid);
-	sip_pvt_unlink(p);
-	__sip_destroy(p);
-	return NULL;
 }
 
 /*! \brief Convert SIP hangup causes to Asterisk hangup causes */