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-rwxr-xr-xcodecs/codec_g723_1.c8
-rwxr-xr-xrtp.c7
2 files changed, 11 insertions, 4 deletions
diff --git a/codecs/codec_g723_1.c b/codecs/codec_g723_1.c
index e8fd71198..ad56bb02e 100755
--- a/codecs/codec_g723_1.c
+++ b/codecs/codec_g723_1.c
@@ -15,9 +15,10 @@
* the GNU General Public License
*/
-#define TYPE_SILENCE 0x2
#define TYPE_HIGH 0x0
#define TYPE_LOW 0x1
+#define TYPE_SILENCE 0x2
+#define TYPE_DONTSEND 0x3
#define TYPE_MASK 0x3
#include <asterisk/translate.h>
@@ -189,7 +190,9 @@ static struct ast_frame *g723tolin_frameout(struct ast_translator_pvt *pvt)
static int g723_len(unsigned char buf)
{
switch(buf & TYPE_MASK) {
- case TYPE_MASK:
+ case TYPE_DONTSEND:
+ return 0;
+ break;
case TYPE_SILENCE:
return 4;
break;
@@ -293,6 +296,7 @@ static struct ast_frame *lintog723_frameout(struct ast_translator_pvt *pvt)
#endif
/* Assume 8000 Hz */
tmp->f.samples += 30;
+ /* FIXME:SLD: Shouldn't the [0] be [cnt]?? */
cnt += g723_len(tmp->outbuf[0]);
tmp->tail -= Frame;
/* Move the data at the end of the buffer to the front */
diff --git a/rtp.c b/rtp.c
index 6f628fee8..aed30a379 100755
--- a/rtp.c
+++ b/rtp.c
@@ -33,9 +33,10 @@
#include <asterisk/acl.h>
#include <asterisk/channel_pvt.h>
-#define TYPE_SILENCE 0x2
#define TYPE_HIGH 0x0
#define TYPE_LOW 0x1
+#define TYPE_SILENCE 0x2
+#define TYPE_DONTSEND 0x3
#define TYPE_MASK 0x3
static int dtmftimeout = 300; /* 300 samples */
@@ -88,7 +89,9 @@ int ast_rtp_fd(struct ast_rtp *rtp)
static int g723_len(unsigned char buf)
{
switch(buf & TYPE_MASK) {
- case TYPE_MASK:
+ case TYPE_DONTSEND:
+ return 0;
+ break;
case TYPE_SILENCE:
return 4;
break;