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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2008-04-21 14:42:03 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2008-04-21 14:42:03 +0000
commitf5d853b157fd24e508eef152d288b9b3649265f1 (patch)
tree5de4b28a93a178c2665d5091b2aa854e686b9076 /channels
parent86d9c6a80221de4fe60333346b6dbdc2635ccd6f (diff)
Merged revisions 114323 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r114323 | file | 2008-04-21 11:40:33 -0300 (Mon, 21 Apr 2008) | 12 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@114324 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 b976e0ebc..edd110412 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5616,7 +5616,7 @@ 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;
}