[asterisk-commits] kpfleming: branch 1.8 r350789 - in /branches/1.8: addons/ channels/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Sat Jan 14 09:23:36 CST 2012


Author: kpfleming
Date: Sat Jan 14 09:23:32 2012
New Revision: 350789

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=350789
Log:
Correct some 'set-but-not-used' variable warnings.


Modified:
    branches/1.8/addons/chan_mobile.c
    branches/1.8/channels/chan_h323.c

Modified: branches/1.8/addons/chan_mobile.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/addons/chan_mobile.c?view=diff&rev=350789&r1=350788&r2=350789
==============================================================================
--- branches/1.8/addons/chan_mobile.c (original)
+++ branches/1.8/addons/chan_mobile.c Sat Jan 14 09:23:32 2012
@@ -2276,14 +2276,13 @@
  */
 static char *hfp_parse_cusd(struct hfp_pvt *hfp, char *buf)
 {
-	int i, state, message_start, message_end;
+	int i, message_start, message_end;
 	char *cusd;
 	size_t s;
 
 	/* parse cusd message in the following format:
 	 * +CUSD: 0,"100,00 EURO, valid till 01.01.2010, you are using tariff "Mega Tariff". More informations *111#."
 	 */
-	state = 0;
 	message_start = 0;
 	message_end = 0;
 	s = strlen(buf);
@@ -2681,7 +2680,7 @@
 {
 	int i, state, group;
 	size_t s;
-	char *indicator = NULL, *values;
+	char *indicator = NULL;
 
 	hfp->nocallsetup = 1;
 
@@ -2720,7 +2719,6 @@
 			}
 			break;
 		case 5: /* mark the start of the value range */
-			values = &buf[i];
 			state++;
 			break;
 		case 6: /* find the end of the value range */
@@ -2992,7 +2990,6 @@
 	sdp_list_t  *l2cap_list = 0, *rfcomm_list = 0, *root_list = 0, *proto_list = 0, *access_proto_list = 0, *svc_uuid_list = 0;
 	sdp_data_t *channel = 0;
 
-	int err = 0;
 	sdp_session_t *session = 0;
 
 	sdp_record_t *record = sdp_record_alloc();
@@ -3029,7 +3026,7 @@
 	if (!(session = sdp_connect(BDADDR_ANY, BDADDR_LOCAL, SDP_RETRY_IF_BUSY)))
 		ast_log(LOG_WARNING, "Failed to connect sdp and create session.\n");
 	else
-		err = sdp_record_register(session, record, 0);
+		sdp_record_register(session, record, 0);
 
 	sdp_data_free(channel);
 	sdp_list_free(rfcomm_list, 0);

Modified: branches/1.8/channels/chan_h323.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_h323.c?view=diff&rev=350789&r1=350788&r2=350789
==============================================================================
--- branches/1.8/channels/chan_h323.c (original)
+++ branches/1.8/channels/chan_h323.c Sat Jan 14 09:23:32 2012
@@ -1757,7 +1757,6 @@
 }
 static struct ast_channel *oh323_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause)
 {
-	format_t oldformat;
 	struct oh323_pvt *pvt;
 	struct ast_channel *tmpc = NULL;
 	char *dest = (char *)data;
@@ -1773,7 +1772,6 @@
 		ast_log(LOG_WARNING, "Unable to build pvt data for '%s'\n", (char *)data);
 		return NULL;
 	}
-	oldformat = format;
 	format &= AST_FORMAT_AUDIO_MASK;
 	if (!format) {
 		ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%s'\n", ast_getformatname_multiple(tmp, sizeof(tmp), format));
@@ -3201,6 +3199,7 @@
 	return res;
 }
 
+#if 0
 static char *convertcap(format_t cap)
 {
 	switch (cap) {
@@ -3227,6 +3226,7 @@
 		return NULL;
 	}
 }
+#endif
 
 static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, format_t codecs, int nat_active)
 {
@@ -3234,13 +3234,18 @@
 	struct oh323_pvt *pvt;
 	struct sockaddr_in them = { 0, };
 	struct sockaddr_in us = { 0, };
+#if 0	/* Native bridge still isn't ready */
 	char *mode;
+#endif
 
 	if (!rtp) {
 		return 0;
 	}
 
+#if 0	/* Native bridge still isn't ready */
 	mode = convertcap(chan->writeformat);
+#endif
+
 	pvt = (struct oh323_pvt *) chan->tech_pvt;
 	if (!pvt) {
 		ast_log(LOG_ERROR, "No Private Structure, this is bad\n");




More information about the asterisk-commits mailing list