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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2008-04-01 17:52:44 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2008-04-01 17:52:44 +0000
commit9d7a9b379a7af046e7d8784ea0ece2eb5a81e144 (patch)
tree47562d417568fd35a6e745b31f97a75213d1dbf0 /channels/chan_sip.c
parentee5da336bb69c9d4b8bb8a21711d801c6769d0d3 (diff)
Merged revisions 112205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r112205 | file | 2008-04-01 14:48:52 -0300 (Tue, 01 Apr 2008) | 12 lines Merged revisions 112204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 lines Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered. (closes issue #11823) Reported by: SDamm ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@112206 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/chan_sip.c')
-rw-r--r--channels/chan_sip.c6
1 files changed, 6 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 52149c955..10ba4cb44 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5607,7 +5607,13 @@ 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) {
+ fr = &ast_null_frame;
+ }
+
sip_pvt_unlock(p);
+
return fr;
}