From: Joshua Colp <jcolp@digium.com>
Date: Tue, 1 Apr 2008 17:48:52 +0000 (+0000)
Subject: Merged revisions 112204 via svnmerge from
X-Git-Tag: 1.6.2.0-beta1~2575
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=a8be22f9da906efca51e68c7e3db461629be9cff;p=thirdparty%2Fasterisk.git

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

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 1982f1fc87..e68718646a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5647,7 +5647,13 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
 		}
 	}
 
+	/* 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;
 }