[asterisk-commits] file: branch 1.6.0 r112206 - in /branches/1.6.0: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Tue Apr  1 12:52:44 2008
New Revision: 112206

URL: http://svn.digium.com/view/asterisk?view=rev&rev=112206
Log:
Merged revisions 112205 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r112205 | file | 2008-04-01 14:48:52 -0300 (Tue, 01 Apr 2008) | 12 lines

Merged revisions 112204 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 lines

Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered.
(closes issue #11823)
Reported by: SDamm

........

................

Modified:
    branches/1.6.0/   (props changed)
    branches/1.6.0/channels/chan_sip.c

Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=112206&r1=112205&r2=112206
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Tue Apr  1 12:52:44 2008
@@ -5607,7 +5607,13 @@
 		}
 	}
 
+	/* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
+	if (p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
+		fr = &ast_null_frame;
+	}
+
 	sip_pvt_unlock(p);
+
 	return fr;
 }