more merge from trunk (comments and change a static function name)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44805 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -11759,7 +11759,7 @@ static int handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_requ
 }
 
 /*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
-static void stop_data_flows(struct sip_pvt *p)
+static void stop_media_flows(struct sip_pvt *p)
 {
 	/* Immediately stop RTP, VRTP and UDPTL as applicable */
 	if (p->rtp)
@@ -11960,7 +11960,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
 					ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
 				ast_set_flag(&p->flags[0], SIP_ALREADYGONE);	
 	
-				stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
+				stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
 
 				/* XXX Locking issues?? XXX */
 				switch(resp) {
@@ -13694,7 +13694,7 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
 			ast_log(LOG_DEBUG, "Got CANCEL on an answered call. Ignoring... \n");
 		return 0;
 	}
-	stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
+	stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
 
 	if (p->owner)
 		ast_queue_hangup(p->owner);
@@ -13743,7 +13743,7 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
 		}
 	}
 
-	stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
+	stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
 
 	if (!ast_strlen_zero(get_header(req, "Also"))) {
 		ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method.  Ask vendor to support REFER instead\n",
@@ -16672,7 +16672,7 @@ static int sip_reload(int fd, int argc, char *argv[])
 	return 0;
 }
 
-/*! \brief  reload: Part of Asterisk module interface */
+/*! \brief Part of Asterisk module interface */
 static int reload(void)
 {
 	return sip_reload(0, 0, NULL);
@@ -16831,7 +16831,7 @@ static struct ast_cli_entry cli_sip[] = {
 	sip_reload_usage },
 };
 
-/*! \brief  load_module: PBX load module - initialization */
+/*! \brief PBX load module - initialization */
 static int load_module(void)
 {
 	ASTOBJ_CONTAINER_INIT(&userl);	/* User object list */