mirror of
https://github.com/asterisk/asterisk.git
synced 2025-09-25 15:08:53 +00:00
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: https://origsvn.digium.com/svn/asterisk/trunk@112205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
@@ -5647,7 +5647,13 @@ 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;
|
||||
}
|
||||
|
||||
|
Reference in New Issue
Block a user