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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2008-04-21 14:39:32 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2008-04-21 14:39:32 +0000
commit39246bc3ad94943e0a1626da66d0d239e7839600 (patch)
treeed555b17071de6bfa89ec22be23a7259d21f4bf6 /channels
parentf71e431b37cfe9fb11571a7d189cb74de422c2eb (diff)
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: http://svn.digium.com/svn/asterisk/branches/1.4@114322 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-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 9e7146a2e..6eeea7e56 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4386,7 +4386,7 @@ 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;
}