[asterisk-commits] igorg: branch 1.8 r377591 - /branches/1.8/channels/chan_unistim.c

[SVN commits to the Asterisk project]

Author: igorg
Date: Mon Dec 10 00:40:18 2012
New Revision: 377591

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=377591
Log:
Fix codec mismatch

Fix code to send in both rx and tx open stream messages correct codecs. Found that on phase 0/1 phones wrong codecs cause to no audio in some situations. 

(issue ASTERISK-20183)


Modified:
    branches/1.8/channels/chan_unistim.c

Modified: branches/1.8/channels/chan_unistim.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_unistim.c?view=diff&rev=377591&r1=377590&r2=377591
==============================================================================
--- branches/1.8/channels/chan_unistim.c (original)
+++ branches/1.8/channels/chan_unistim.c Mon Dec 10 00:40:18 2012
@@ -2151,9 +2151,9 @@
 			buffsend[16] = (htons(sin.sin_port) & 0x00ff);
 			buffsend[20] = (us.sin_port & 0xff00) >> 8;
 			buffsend[19] = (us.sin_port & 0x00ff);
-			buffsend[11] = codec;
-		}
-		buffsend[12] = codec;
+		}
+		buffsend[11] = codec; /* rx */
+		buffsend[12] = codec; /* tx */
 		send_client(SIZE_HEADER + sizeof(packet_send_open_audio_stream_tx), buffsend,
 				   sub->parent->parent->session);
 
@@ -2181,9 +2181,9 @@
 			buffsend[16] = (htons(sin.sin_port) & 0x00ff);
 			buffsend[20] = (us.sin_port & 0xff00) >> 8;
 			buffsend[19] = (us.sin_port & 0x00ff);
-			buffsend[12] = codec;
-		}
-		buffsend[11] = codec;
+		}
+		buffsend[11] = codec; /* rx */
+		buffsend[12] = codec; /* tx */
 		send_client(SIZE_HEADER + sizeof(packet_send_open_audio_stream_rx), buffsend,
 				   sub->parent->parent->session);
 	} else {