[asterisk-commits] dvossel: branch dvossel/hd_conferencing_ftw r307968 - /team/dvossel/hd_confer...

[SVN commits to the Asterisk project]

Author: dvossel
Date: Tue Feb 15 16:42:22 2011
New Revision: 307968

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=307968
Log:
Ability to use MUTEAUDIO dialplan function in one direction without it downsampling the other direction

Modified:
    team/dvossel/hd_conferencing_ftw/res/res_mutestream.c

Modified: team/dvossel/hd_conferencing_ftw/res/res_mutestream.c
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/hd_conferencing_ftw/res/res_mutestream.c?view=diff&rev=307968&r1=307967&r2=307968
==============================================================================
--- team/dvossel/hd_conferencing_ftw/res/res_mutestream.c (original)
+++ team/dvossel/hd_conferencing_ftw/res/res_mutestream.c Tue Feb 15 16:42:22 2011
@@ -173,7 +173,7 @@
 		ast_datastore_free(datastore);
 		return NULL;
 	}
-	ast_audiohook_init(&mute->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Mute", 0);
+	ast_audiohook_init(&mute->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Mute", AST_AUDIOHOOK_NATIVE_SLIN_CAPABLE);
 	mute->audiohook.manipulate_callback = mute_callback;
 	datastore->data = mute;
 	return datastore;