Misc minor changes in chan_sip.

* Add load failure exit if primary SIP container(s) could not get created
in chan_sip.c:load_module().

* Removed a redundant static prototype.

* Some typos.

* Some whitespace.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325935 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5682d4c658dd19be1264e7c14ae31fd338a264f0..077d84efe5c39a490286ca5f9b14ceaf7484d9c7 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1319,7 +1319,6 @@ static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, con
 
 /*--- Misc functions */
 static void check_rtp_timeout(struct sip_pvt *dialog, time_t t);
-static int sip_do_reload(enum channelreloadreason reason);
 static int reload_config(enum channelreloadreason reason);
 static int expire_register(const void *data);
 static void *do_monitor(void *data);
@@ -13088,7 +13087,7 @@ static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xm
        return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
 }
 
-/*! \brief return the request and response heade for a 401 or 407 code */
+/*! \brief return the request and response header for a 401 or 407 code */
 static void auth_headers(enum sip_auth_type code, char **header, char **respheader)
 {
        if (code == WWW_AUTH) {                 /* 401 */
@@ -16892,13 +16891,13 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
                ast_cli(fd, "  Status       : ");
                peer_status(peer, status, sizeof(status));
                ast_cli(fd, "%s\n", status);
-               ast_cli(fd, "  Useragent    : %s\n", peer->useragent);
-               ast_cli(fd, "  Reg. Contact : %s\n", peer->fullcontact);
+               ast_cli(fd, "  Useragent    : %s\n", peer->useragent);
+               ast_cli(fd, "  Reg. Contact : %s\n", peer->fullcontact);
                ast_cli(fd, "  Qualify Freq : %d ms\n", peer->qualifyfreq);
                if (peer->chanvars) {
-                       ast_cli(fd, "  Variables    :\n");
+                       ast_cli(fd, "  Variables    :\n");
                        for (v = peer->chanvars ; v ; v = v->next)
-                               ast_cli(fd, "                 %s = %s\n", v->name, v->value);
+                               ast_cli(fd, "                 %s = %s\n", v->name, v->value);
                }
 
                ast_cli(fd, "  Sess-Timers  : %s\n", stmode2str(peer->stimer.st_mode_oper));
@@ -16992,13 +16991,13 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
                astman_append(s, "Status: ");
                peer_status(peer, status, sizeof(status));
                astman_append(s, "%s\r\n", status);
-               astman_append(s, "SIP-Useragent: %s\r\n", peer->useragent);
-               astman_append(s, "Reg-Contact: %s\r\n", peer->fullcontact);
+               astman_append(s, "SIP-Useragent: %s\r\n", peer->useragent);
+               astman_append(s, "Reg-Contact: %s\r\n", peer->fullcontact);
                astman_append(s, "QualifyFreq: %d ms\r\n", peer->qualifyfreq);
                astman_append(s, "Parkinglot: %s\r\n", peer->parkinglot);
                if (peer->chanvars) {
                        for (v = peer->chanvars ; v ; v = v->next) {
-                               astman_append(s, "ChanVariable: %s=%s\r\n", v->name, v->value);
+                               astman_append(s, "ChanVariable: %s=%s\r\n", v->name, v->value);
                        }
                }
                astman_append(s, "SIP-Use-Reason-Header : %s\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)) ? "Y" : "N");
@@ -18596,7 +18595,7 @@ static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int d
                username = p->authname;
                secret = p->relatedpeer 
                        && !ast_strlen_zero(p->relatedpeer->remotesecret)
-                       ? p->relatedpeer->remotesecret : p->peersecret;
+                               ? p->relatedpeer->remotesecret : p->peersecret;
                md5secret = p->peermd5secret;
        }
        if (ast_strlen_zero(username))  /* We have no authentication */
@@ -18620,7 +18619,7 @@ static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int d
 
        /* only include the opaque string if it's set */
        if (!ast_strlen_zero(p->opaque)) {
-         snprintf(opaque, sizeof(opaque), ", opaque=\"%s\"", p->opaque);
+               snprintf(opaque, sizeof(opaque), ", opaque=\"%s\"", p->opaque);
        }
 
        /* XXX We hard code our qop to "auth" for now.  XXX */
@@ -29308,13 +29307,18 @@ static const struct ast_data_entry sip_data_providers[] = {
 static int load_module(void)
 {
        ast_verbose("SIP channel loading...\n");
+
        /* the fact that ao2_containers can't resize automatically is a major worry! */
        /* if the number of objects gets above MAX_XXX_BUCKETS, things will slow down */
        peers = ao2_t_container_alloc(HASH_PEER_SIZE, peer_hash_cb, peer_cmp_cb, "allocate peers");
        peers_by_ip = ao2_t_container_alloc(HASH_PEER_SIZE, peer_iphash_cb, peer_ipcmp_cb, "allocate peers_by_ip");
        dialogs = ao2_t_container_alloc(HASH_DIALOG_SIZE, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs");
        threadt = ao2_t_container_alloc(HASH_DIALOG_SIZE, threadt_hash_cb, threadt_cmp_cb, "allocate threadt table");
-       
+       if (!peers || !peers_by_ip || !dialogs || !threadt) {
+               ast_log(LOG_ERROR, "Unable to create primary SIP container(s)\n");
+               return AST_MODULE_LOAD_FAILURE;
+       }
+
        ASTOBJ_CONTAINER_INIT(&regl); /* Registry object list -- not searched for anything */
        ASTOBJ_CONTAINER_INIT(&submwil); /* MWI subscription object list */
 
@@ -29332,7 +29336,7 @@ static int load_module(void)
        sip_reloadreason = CHANNEL_MODULE_LOAD;
 
        can_parse_xml = sip_is_xml_parsable();
-       if(reload_config(sip_reloadreason)) {   /* Load the configuration from sip.conf */
+       if (reload_config(sip_reloadreason)) {  /* Load the configuration from sip.conf */
                return AST_MODULE_LOAD_DECLINE;
        }
 

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 400351c0a59eb95abdf547615926663a07b93c9e..d2712871d70f762d1f6b6392f4c8e3946d710392 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -130,7 +130,7 @@ allowoverlap=no                 ; Disable overlap dialing support. (Default is y
                                 ; asterisk.conf, it defaults to that system name
                                 ; Realms MUST be globally unique according to RFC 3261
                                 ; Set this to your host name or domain name
-;domainsasrealm=no              ; Use domans list as realms
+;domainsasrealm=no              ; Use domains list as realms
                                 ; You can serve multiple Realms specifying several
                                 ; 'domain=...' directives (see below). 
                                 ; In this case Realm will be based on request 'From'/'To' header