Version 0.3.0 from FTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@586 65c4cc65-6c06-0410-ace0-fbb531ad65f3

configs/sip.conf.sample

@@ -5,6 +5,8 @@
 port = 5060			; Port to bind to
 bindaddr = 0.0.0.0		; Address to bind to
 context = default		; Default for incoming calls
+;tos=lowdelay
+;tos=184
 
 ;[snomsip]
 ;type=friend
@@ -25,6 +27,8 @@ context = default		; Default for incoming calls
 ;username=cisco
 ;secret=blah
 ;host=dynamic
+;canreinvite=no			; Cisco poops on reinvite sometimes
+;qualify=200			; Qualify peer is no more than 200ms away
 ;defaultip=192.168.0.4
 
 ;[cisco1]

include/asterisk/rtp.h

@@ -32,6 +32,8 @@ struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io);
 
 void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
 
+void ast_rtp_get_peer(struct ast_rtp *rpt, struct sockaddr_in *them);
+
 void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
 
 void ast_rtp_destroy(struct ast_rtp *rtp);
@@ -42,6 +44,10 @@ void ast_rtp_set_data(struct ast_rtp *rtp, void *data);
 
 int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *f);
 
+int ast_rtp_senddigit(struct ast_rtp *rtp, char digit);
+
+int ast_rtp_settos(struct ast_rtp *rtp, int tos);
+
 int ast2rtp(int id);
 
 int rtp2ast(int id);