Minor configuration fixes/standardizations (bug #3317)

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@4761 f38db490-d61c-443f-a65b-d21fe96a405b

2 files changed, 10 insertions, 9 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d2ed0ddeb..292ddc5cd 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -164,7 +164,7 @@ static int rpeerobjs = 0;
 static int apeerobjs = 0;
 static int regobjs = 0;
 
-static int global_allowguest = 0;    /* allow unathuncated peers to connect? */
+static int global_allowguest = 0;    /* allow unauthenticated users/peers to connect? */
 
 #define DEFAULT_MWITIME 10
 static int global_mwitime = DEFAULT_MWITIME;	/* Time between MWI checks for peers */
@@ -209,7 +209,8 @@ static int compactheaders = 0; 						/* send compact sip headers */
 static int recordhistory = 0;				/* Record SIP history. Off by default */
 
 static char global_musicclass[MAX_LANGUAGE] = "";	/* Global music on hold class */
-static char global_realm[AST_MAX_EXTENSION] = "asterisk"; 	/* Default realm */
+#define DEFAULT_REALM	"asterisk"
+static char global_realm[AST_MAX_EXTENSION] = DEFAULT_REALM; 	/* Default realm */
 static char regcontext[AST_MAX_EXTENSION] = "";		/* Context for auto-extensions */
 
 /* Expire slowly */
@@ -285,7 +286,7 @@ struct sip_history {
 #define SIP_NAT_ALWAYS		(3 << 18)
 /* re-INVITE related settings */
 #define SIP_REINVITE		(3 << 20)	/* two bits used */
-#define SIP_CAN_REINVITE	(1 << 20)	/* allow peers to be reinvited to send media directly to us */
+#define SIP_CAN_REINVITE	(1 << 20)	/* allow peers to be reinvited to send media directly p2p */
 #define SIP_REINVITE_UPDATE	(2 << 20)	/* use UPDATE (RFC3311) when reinviting this peer */
 /* "insecure" settings */
 #define SIP_INSECURE		(3 << 22)	/* three settings, uses two bits */
@@ -8890,7 +8891,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int
 			} else if (!strcasecmp(v->name, "mask")) {
 				maskfound++;
 				inet_aton(v->value, &peer->mask);
-			} else if (!strcasecmp(v->name, "port")) {
+			} else if (!strcasecmp(v->name, "port") || !strcasecmp(v-name, "bindport") {
 				if (!realtime && ast_test_flag(peer, SIP_DYNAMIC))
 					peer->defaddr.sin_port = htons(atoi(v->value));
 				else
@@ -9011,7 +9012,7 @@ static int reload_config(void)
 	externrefresh = 10;
 	strncpy(default_useragent, DEFAULT_USERAGENT, sizeof(default_useragent) - 1);
 	strncpy(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime) - 1);
-	global_realm[sizeof(global_realm)-1] = '\0';
+	strncpy(global_realm, DEFAULT_REALM, sizeof(global_realm) - 1);
 	strncpy(global_musicclass, "default", sizeof(global_musicclass) - 1);
 	strncpy(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid) - 1);
 	memset(&outboundproxyip, 0, sizeof(outboundproxyip));
@@ -9085,7 +9086,7 @@ static int reload_config(void)
 			compactheaders = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "notifymimetype")) {
 			strncpy(default_notifymime, v->value, sizeof(default_notifymime) - 1);
-		} else if (!strcasecmp(v->name, "musicclass")) {
+		} else if (!strcasecmp(v->name, "musicclass") || !strcasecmp(v->name, "musiconhold") {
 			strncpy(global_musicclass, v->value, sizeof(global_musicclass) - 1);
 		} else if (!strcasecmp(v->name, "language")) {
 			strncpy(default_language, v->value, sizeof(default_language)-1);
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index a3585c2d7..7ba7b53fc 100755
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -24,13 +24,11 @@
 context=default			; Default context for incoming calls
 ;allowguest=no			; Allow or reject guest calls (default is yes, this can also be set to 'osp'
 				; if asterisk was compiled with OSP support.
-;recordhistory=yes		; Record SIP history by default 
-				; (see sip history / sip no history)
 ;realm=mydomain.tld		; Realm for digest authentication
 				; defaults to "asterisk"
 				; Realms MUST be globally unique according to RFC 3261
 				; Set this to your host name or domain name
-port=5060			; UDP Port to bind to (SIP standard port is 5060)
+bindport=5060			; UDP Port to bind to (SIP standard port is 5060)
 bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
 srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 				; Note: Asterisk only uses the first host 
@@ -49,6 +47,8 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ;notifymimetype=text/plain	; Allow overriding of mime type in MWI NOTIFY
 ;checkmwi=10			; Default time between mailbox checks for peers
 ;videosupport=yes		; Turn on support for SIP video
+;recordhistory=yes		; Record SIP history by default 
+				; (see sip history / sip no history)
 
 ;disallow=all			; First disallow all codecs
 ;allow=ulaw			; Allow codecs in order of preference