From a79214b5b1fb27d45f599166953f622bfcb0dc3e Mon Sep 17 00:00:00 2001
From: Joshua Colp <jcolp@digium.com>
Date: Mon, 21 Apr 2008 14:40:33 +0000
Subject: [PATCH] Merged revisions 114322 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines
Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call.
(closes issue #12440)
Reported by: aragon
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 8a583f718e..79fc483990 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5880,7 +5880,7 @@ 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) {
+ if (fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
fr = &ast_null_frame;
}
--
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