[asterisk-commits] mmichelson: branch 1.6.0 r130795 - in /branches/1.6.0: ./ apps/app_dial.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jul 14 12:55:01 CDT 2008
Author: mmichelson
Date: Mon Jul 14 12:55:01 2008
New Revision: 130795
URL: http://svn.digium.com/view/asterisk?view=rev&rev=130795
Log:
Merged revisions 130794 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul 2008) | 16 lines
Merged revisions 130792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines
Add a check to the CAN_EARLY_BRIDGE macro in app_dial to
be sure there are no audiohooks present on the channels
involved. This fixed a one-way audio situation I had in
my test setup. I couldn't find any open issues that suggested
one-way audio with regards to mixmonitor (or other audiohook)
usage, though.
........
................
Modified:
branches/1.6.0/ (props changed)
branches/1.6.0/apps/app_dial.c
Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.0/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/apps/app_dial.c?view=diff&rev=130795&r1=130794&r2=130795
==============================================================================
--- branches/1.6.0/apps/app_dial.c (original)
+++ branches/1.6.0/apps/app_dial.c Mon Jul 14 12:55:01 2008
@@ -321,9 +321,10 @@
AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
END_OPTIONS );
-#define CAN_EARLY_BRIDGE(flags) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
+#define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
- OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK))
+ OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK) && \
+ !chan->audiohooks && !peer->audiohooks)
/*
* The list of active channels
@@ -671,7 +672,7 @@
DIAL_NOFORWARDHTML);
ast_copy_string(c->dialcontext, "", sizeof(c->dialcontext));
ast_copy_string(c->exten, "", sizeof(c->exten));
- if (CAN_EARLY_BRIDGE(peerflags))
+ if (CAN_EARLY_BRIDGE(peerflags, in, peer))
/* Setup early bridge if appropriate */
ast_channel_early_bridge(in, peer);
}
@@ -698,7 +699,7 @@
case AST_CONTROL_RINGING:
ast_verb(3, "%s is ringing\n", c->name);
/* Setup early media if appropriate */
- if (single && CAN_EARLY_BRIDGE(peerflags))
+ if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
ast_channel_early_bridge(in, c);
if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK)) {
ast_indicate(in, AST_CONTROL_RINGING);
@@ -708,7 +709,7 @@
case AST_CONTROL_PROGRESS:
ast_verb(3, "%s is making progress passing it to %s\n", c->name, in->name);
/* Setup early media if appropriate */
- if (single && CAN_EARLY_BRIDGE(peerflags))
+ if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
ast_channel_early_bridge(in, c);
if (!ast_test_flag64(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROGRESS);
@@ -723,7 +724,7 @@
break;
case AST_CONTROL_PROCEEDING:
ast_verb(3, "%s is proceeding passing it to %s\n", c->name, in->name);
- if (single && CAN_EARLY_BRIDGE(peerflags))
+ if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
ast_channel_early_bridge(in, c);
if (!ast_test_flag64(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROCEEDING);
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