[asterisk-commits] file: branch 1.4 r64754 -
/branches/1.4/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Thu May 17 09:10:13 MST 2007
Author: file
Date: Thu May 17 11:10:12 2007
New Revision: 64754
URL: http://svn.digium.com/view/asterisk?view=rev&rev=64754
Log:
Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu)
Modified:
branches/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=64754&r1=64753&r2=64754
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Thu May 17 11:10:12 2007
@@ -17024,6 +17024,11 @@
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))
More information about the asterisk-commits
mailing list