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Rename ast_rtp_early_media to ast_rtp_early_bridge to avoid confusion.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2006-06-09 09:47:44 +00:00
parent ce10b34ca6
commit 9f5aa13142
3 changed files with 10 additions and 8 deletions
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12
apps/app_dial.c
View File

@@ -576,8 +576,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
OPT_CALLEE_PARK | OPT_CALLER_PARK |
DIAL_NOFORWARDHTML);
/* Setup early media if appropriate */
ast_rtp_early_media(in, peer);
/* Setup RTP early bridge if appropriate */
ast_rtp_early_bridge(in, peer);
}
/* If call has been answered, then the eventual hangup is likely to be normal hangup */
in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
@@ -606,7 +606,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", c->name);
/* Setup early media if appropriate */
if (single)
ast_rtp_early_media(in, c);
ast_rtp_early_bridge(in, c);
if (!(*sentringing) && !ast_test_flag(outgoing, OPT_MUSICBACK)) {
ast_indicate(in, AST_CONTROL_RINGING);
(*sentringing)++;
@@ -617,7 +617,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
ast_verbose (VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", c->name, in->name);
/* Setup early media if appropriate */
if (single)
ast_rtp_early_media(in, c);
ast_rtp_early_bridge(in, c);
if (!ast_test_flag(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROGRESS);
break;
@@ -630,7 +630,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
if (option_verbose > 2)
ast_verbose (VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", c->name, in->name);
if (single)
ast_rtp_early_media(in, c);
ast_rtp_early_bridge(in, c);
if (!ast_test_flag(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROCEEDING);
break;
@@ -1608,7 +1608,7 @@ out:
sentringing = 0;
ast_indicate(chan, -1);
}
ast_rtp_early_media(chan, NULL);
ast_rtp_early_bridge(chan, NULL);
hanguptree(outgoing, NULL);
pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
if (option_debug)

4
include/asterisk/rtp.h
View File

@@ -182,7 +182,9 @@ void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
int ast_rtp_early_media(struct ast_channel *dest, struct ast_channel *src);
/*! \brief If possible, create an early bridge directly between the devices without
having to send a re-invite later */
int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src);
void ast_rtp_stop(struct ast_rtp *rtp);

2
rtp.c
View File

@@ -1274,7 +1274,7 @@ static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
return cur;
}
int ast_rtp_early_media(struct ast_channel *dest, struct ast_channel *src)
int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src)
{
struct ast_rtp *destp, *srcp=NULL; /* Audio RTP Channels */
struct ast_rtp *vdestp, *vsrcp=NULL; /* Video RTP channels */