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authorKevin P. Fleming <kpfleming@digium.com>2006-05-16 22:11:02 +0000
committerKevin P. Fleming <kpfleming@digium.com>2006-05-16 22:11:02 +0000
commitc3dce57d95b4f313dde62f4d607a30f5c101596c (patch)
tree16a03f4e88229adb3de2cd5470afd1e4048a3956 /include
parentca8f9083c0142f6c5be55408a80487dd44f2ea3c (diff)
mark RTP sessions that are not carrying DTMF
allow native bridging of RTP sessions that are not carrying DTMF even when the bridge needs to listen to DTMF (when SIP INFO is used for DTMF, for example) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'include')
-rw-r--r--include/asterisk/rtp.h3
1 files changed, 3 insertions, 0 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 7235639f1..5b03d1d86 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -151,6 +151,9 @@ char *ast_rtp_lookup_mime_multiple(char *buf, int size, const int capability, co
void ast_rtp_setnat(struct ast_rtp *rtp, int nat);
+/*! \brief Indicate whether this RTP session is carrying DTMF or not */
+void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf);
+
int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
int ast_rtp_proto_register(struct ast_rtp_protocol *proto);