Merged revisions 114322 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114323 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 8a583f718eb15569bb79e51dc3085efdc44a2d30..79fc483990b98cc616bd1f87362a3ab08a43a8fa 100644 (file)
--- 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;
        }