[Asterisk-cvs] asterisk/channels chan_sip.c,1.204,1.205

martinp at lists.digium.com martinp at lists.digium.com
Mon Nov 3 18:08:08 CST 2003


Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv11833/channels

Modified Files:
	chan_sip.c 
Log Message:
Make SIP_CODEC warn about not supported/not configured codec


Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.204
retrieving revision 1.205
diff -u -d -r1.204 -r1.205
--- chan_sip.c	31 Oct 2003 16:13:49 -0000	1.204
+++ chan_sip.c	4 Nov 2003 00:34:17 -0000	1.205
@@ -1128,11 +1128,11 @@
 	
 		codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC");
 		if (codec) {
-			ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
 			fmt=ast_getformatbyname(codec);
 			if (fmt) {
+				ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
 				p->capability=fmt;
-			} else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized codec: %s\n",codec);
+			} else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
 		}
 
 		ast_setstate(ast, AST_STATE_UP);




More information about the svn-commits mailing list