Minor configuration fixes/standardizations (bug #3317)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4761 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d2ed0ddeb429a6ba51c5a95e0f9609873e0312ae..292ddc5cd48d2976d45dd71ba126d9d01d2321df 100755 (executable)
--- 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 a3585c2d764808e6b53c0f65138256a04c0211d9..7ba7b53fccf17bb4f9517378762a3ab4b4829266 100755 (executable)
--- 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