Merged revisions 285710 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.6.2

........
  r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines
  
  Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
  
  (closes issue #16903)
  Reported by: Nick_Lewis
  Patches: 
        pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
  Tested by: Nick_Lewis
........


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 21 insertions, 0 deletions

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 396aed533..6ddd33902 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -270,6 +270,7 @@ static int ast_rtp_get_stat(struct ast_rtp_instance *instance, struct ast_rtp_in
 static int ast_rtp_dtmf_compatible(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1);
 static void ast_rtp_stun_request(struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username);
 static void ast_rtp_stop(struct ast_rtp_instance *instance);
+static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos, const char* desc);
 
 /* RTP Engine Declaration */
 static struct ast_rtp_engine asterisk_rtp_engine = {
@@ -293,6 +294,7 @@ static struct ast_rtp_engine asterisk_rtp_engine = {
 	.dtmf_compatible = ast_rtp_dtmf_compatible,
 	.stun_request = ast_rtp_stun_request,
 	.stop = ast_rtp_stop,
+	.qos = ast_rtp_qos_set,
 };
 
 static inline int rtp_debug_test_addr(struct ast_sockaddr *addr)
@@ -715,12 +717,24 @@ static void ast_rtp_update_source(struct ast_rtp_instance *instance)
 static void ast_rtp_change_source(struct ast_rtp_instance *instance)
 {
 	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+	struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance);
 	unsigned int ssrc = ast_random();
 
+	if (!rtp->lastts) {
+		ast_debug(3, "Not changing SSRC since we haven't sent any RTP yet\n");
+		return;
+	}
+
 	/* We simply set this bit so that the next packet sent will have the marker bit turned on */
 	ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
 
 	ast_debug(3, "Changing ssrc from %u to %u due to a source change\n", rtp->ssrc, ssrc);
+
+	if (srtp) {
+		ast_debug(3, "Changing ssrc for SRTP from %u to %u\n", rtp->ssrc, ssrc);
+		res_srtp->change_source(srtp, rtp->ssrc, ssrc);
+	}
+
 	rtp->ssrc = ssrc;
 
 	return;
@@ -2549,6 +2563,13 @@ static void ast_rtp_stop(struct ast_rtp_instance *instance)
 	ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
 }
 
+static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
+{
+	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+	return ast_set_qos(rtp->s, tos, cos, desc);
+}
+
 static char *rtp_do_debug_ip(struct ast_cli_args *a)
 {
 	char *arg = ast_strdupa(a->argv[4]);