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Rename ast_rtp_early_media to ast_rtp_early_bridge to avoid confusion.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -576,8 +576,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
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OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
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OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
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OPT_CALLEE_PARK | OPT_CALLER_PARK |
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OPT_CALLEE_PARK | OPT_CALLER_PARK |
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DIAL_NOFORWARDHTML);
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DIAL_NOFORWARDHTML);
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/* Setup early media if appropriate */
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/* Setup RTP early bridge if appropriate */
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ast_rtp_early_media(in, peer);
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ast_rtp_early_bridge(in, peer);
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}
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}
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/* If call has been answered, then the eventual hangup is likely to be normal hangup */
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/* If call has been answered, then the eventual hangup is likely to be normal hangup */
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in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
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in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
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@@ -606,7 +606,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
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ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", c->name);
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ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", c->name);
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/* Setup early media if appropriate */
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/* Setup early media if appropriate */
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if (single)
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if (single)
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ast_rtp_early_media(in, c);
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ast_rtp_early_bridge(in, c);
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if (!(*sentringing) && !ast_test_flag(outgoing, OPT_MUSICBACK)) {
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if (!(*sentringing) && !ast_test_flag(outgoing, OPT_MUSICBACK)) {
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ast_indicate(in, AST_CONTROL_RINGING);
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ast_indicate(in, AST_CONTROL_RINGING);
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(*sentringing)++;
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(*sentringing)++;
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@@ -617,7 +617,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
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ast_verbose (VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", c->name, in->name);
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ast_verbose (VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", c->name, in->name);
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/* Setup early media if appropriate */
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/* Setup early media if appropriate */
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if (single)
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if (single)
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ast_rtp_early_media(in, c);
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ast_rtp_early_bridge(in, c);
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if (!ast_test_flag(outgoing, OPT_RINGBACK))
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if (!ast_test_flag(outgoing, OPT_RINGBACK))
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ast_indicate(in, AST_CONTROL_PROGRESS);
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ast_indicate(in, AST_CONTROL_PROGRESS);
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break;
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break;
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@@ -630,7 +630,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
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if (option_verbose > 2)
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if (option_verbose > 2)
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ast_verbose (VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", c->name, in->name);
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ast_verbose (VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", c->name, in->name);
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if (single)
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if (single)
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ast_rtp_early_media(in, c);
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ast_rtp_early_bridge(in, c);
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if (!ast_test_flag(outgoing, OPT_RINGBACK))
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if (!ast_test_flag(outgoing, OPT_RINGBACK))
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ast_indicate(in, AST_CONTROL_PROCEEDING);
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ast_indicate(in, AST_CONTROL_PROCEEDING);
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break;
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break;
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@@ -1608,7 +1608,7 @@ out:
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sentringing = 0;
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sentringing = 0;
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ast_indicate(chan, -1);
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ast_indicate(chan, -1);
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}
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}
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ast_rtp_early_media(chan, NULL);
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ast_rtp_early_bridge(chan, NULL);
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hanguptree(outgoing, NULL);
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hanguptree(outgoing, NULL);
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pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
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pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
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if (option_debug)
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if (option_debug)
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@@ -182,7 +182,9 @@ void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
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int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
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int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
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int ast_rtp_early_media(struct ast_channel *dest, struct ast_channel *src);
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/*! \brief If possible, create an early bridge directly between the devices without
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having to send a re-invite later */
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int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src);
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void ast_rtp_stop(struct ast_rtp *rtp);
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void ast_rtp_stop(struct ast_rtp *rtp);
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2
rtp.c
2
rtp.c
@@ -1274,7 +1274,7 @@ static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
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return cur;
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return cur;
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}
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}
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int ast_rtp_early_media(struct ast_channel *dest, struct ast_channel *src)
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int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src)
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{
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{
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struct ast_rtp *destp, *srcp=NULL; /* Audio RTP Channels */
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struct ast_rtp *destp, *srcp=NULL; /* Audio RTP Channels */
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struct ast_rtp *vdestp, *vsrcp=NULL; /* Video RTP channels */
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struct ast_rtp *vdestp, *vsrcp=NULL; /* Video RTP channels */
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