[Asterisk-cvs] asterisk/channels chan_sip.c,1.609,1.610

[markster at lists.digium.com]

Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv353/channels

Modified Files:
	chan_sip.c 
Log Message:
Merge OEJ's print groups feature (bug #3228, with changes)


Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.609
retrieving revision 1.610
diff -u -d -r1.609 -r1.610
--- chan_sip.c	2 Jan 2005 00:48:11 -0000	1.609
+++ chan_sip.c	3 Jan 2005 00:51:57 -0000	1.610
@@ -5902,20 +5902,8 @@
 /*--- print_group: Print call group and pickup group ---*/
 static void  print_group(int fd, unsigned int group) 
 {
-	unsigned int i;
-	int first=1;
-
-	for (i=0; i<=31; i++) {	/* Max group is 31 */
-		if (group & (1 << i)) {
-	   		if (!first) {
-				ast_cli(fd, ", ");
-			} else {
-				first=0;
-	  		}
-			ast_cli(fd, "%u", i);
-		}
-    	}
-	ast_cli(fd, " (%u)\n", group);
+	char buf[256];
+	ast_cli(fd, ast_print_group(buf, sizeof(buf), group) );
 }
 
 static const char *dtmfmode2str(int mode)
@@ -5928,6 +5916,7 @@
 	case SIP_DTMF_INBAND:
 		return "inband";
 	}
+	return "<error>";
 }
 
 static const char *insecure2str(int mode)
@@ -5940,6 +5929,7 @@
 	case SIP_INSECURE_VERY:
 		return "very";
 	}
+	return "<error>";
 }
 
 /*--- sip_show_peer: Show one peer in detail ---*/
@@ -9065,6 +9055,7 @@
 	return 0;
 }
 
+/*--- sip_get_rtp_peer: Returns null if we can't reinvite */
 static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan)
 {
 	struct sip_pvt *p;
@@ -9287,12 +9278,14 @@
 	return 0;
 }
 
+/*--- sip_get_codec: Return peers codec ---*/
 static int sip_get_codec(struct ast_channel *chan)
 {
 	struct sip_pvt *p = chan->pvt->pvt;
 	return p->peercapability;	
 }
 
+/*--- sip_rtp: Interface structure with callbacks used to connect to rtp module --*/
 static struct ast_rtp_protocol sip_rtp = {
 	get_rtp_info: sip_get_rtp_peer,
 	get_vrtp_info: sip_get_vrtp_peer,