[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))
    
    
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