Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep...

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64754 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 34bccdee099c7d462efc97204d0ddb95bd81edfd..6e76977876af5dc32dad1628249eb2fb862210d4 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -17024,6 +17024,11 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
                p->redircodecs = codecs;
                changed = 1;
        }
+       if ((p->capability & codecs) != p->capability) {
+               p->jointcapability &= codecs;
+               p->capability &= codecs;
+               changed = 1;
+       }
        if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
                if (chan->_state != AST_STATE_UP) {     /* We are in early state */
                        if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))