From 63de8343958b91c8836c5e6ddf1c0106b40e9fe6 Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Thu, 2 Apr 2009 17:20:52 +0000 Subject: Merge in the RTP engine API. This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- apps/app_dial.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'apps/app_dial.c') diff --git a/apps/app_dial.c b/apps/app_dial.c index 8f6a49ba3..96bb57081 100644 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -54,7 +54,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/utils.h" #include "asterisk/app.h" #include "asterisk/causes.h" -#include "asterisk/rtp.h" +#include "asterisk/rtp_engine.h" #include "asterisk/cdr.h" #include "asterisk/manager.h" #include "asterisk/privacy.h" @@ -745,7 +745,9 @@ static void do_forward(struct chanlist *o, char *new_cid_num, *new_cid_name; struct ast_channel *src; - ast_rtp_make_compatible(c, in, single); + if (single) { + ast_rtp_instance_early_bridge_make_compatible(c, in); + } if (ast_test_flag64(o, OPT_FORCECLID)) { new_cid_num = ast_strdup(S_OR(in->macroexten, in->exten)); new_cid_name = NULL; /* XXX no name ? */ @@ -1745,7 +1747,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst); /* Setup outgoing SDP to match incoming one */ - ast_rtp_make_compatible(tc, chan, !outgoing && !rest); + if (!outgoing && !rest) { + ast_rtp_instance_early_bridge_make_compatible(tc, chan); + } /* Inherit specially named variables from parent channel */ ast_channel_inherit_variables(chan, tc); -- cgit v1.2.3