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authorJoshua Colp <jcolp@digium.com>2008-04-21 14:40:33 +0000
committerJoshua Colp <jcolp@digium.com>2008-04-21 14:40:33 +0000
commita79214b5b1fb27d45f599166953f622bfcb0dc3e (patch)
tree0c9cffe2605ad09862845078623bb33f9a9cd2f0
parent6d5290a85b51a6409c7f94bc09c7b110a9cf137c (diff)
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
-rw-r--r--channels/chan_sip.c2
1 files changed, 1 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 8a583f718..79fc48399 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;
}