From 2dfca4e149fa00bb747b85a380fa2e2fb83737c8 Mon Sep 17 00:00:00 2001
From: Joshua Colp <jcolp@digium.com>
Date: Tue, 1 Apr 2008 17:52:44 +0000
Subject: [PATCH] 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

........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@112206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 channels/chan_sip.c | 6 ++++++
 1 file changed, 6 insertions(+)

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 52149c9556..10ba4cb441 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5607,7 +5607,13 @@ 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;
 }