[Asterisk-cvs] asterisk/channels chan_sip.c,1.183,1.184

markster at lists.digium.com markster at lists.digium.com
Tue Sep 30 18:02:46 CDT 2003


Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv4661/channels

Modified Files:
	chan_sip.c 
Log Message:
Add sayunixtime, chan_sip updates for codec negotiation


Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.183
retrieving revision 1.184
diff -u -d -r1.183 -r1.184
--- chan_sip.c	30 Sep 2003 04:57:20 -0000	1.183
+++ chan_sip.c	30 Sep 2003 23:03:57 -0000	1.184
@@ -2423,7 +2423,7 @@
 	/* Start by sending our preferred codecs */
 	cur = prefs;
 	while(cur) {
-		if (p->capability & cur->codec) {
+		if (p->jointcapability & cur->codec) {
 			if (sipdebug)
 				ast_verbose("Answering with preferred capability %d\n", cur->codec);
 			codec = ast_rtp_lookup_code(p->rtp, 1, cur->codec);
@@ -2445,7 +2445,7 @@
 	}
 	/* Now send any other common codecs, and non-codec formats: */
 	for (x = 1; x <= AST_FORMAT_MAX_AUDIO; x <<= 1) {
-		if ((p->capability & x) && !(alreadysent & x)) {
+		if ((p->jointcapability & x) && !(alreadysent & x)) {
 			if (sipdebug)
 				ast_verbose("Answering with capability %d\n", x);	
 			codec = ast_rtp_lookup_code(p->rtp, 1, x);




More information about the svn-commits mailing list