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Misc minor changes in chan_sip.

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* 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
This commit is contained in:
Richard Mudgett
2011-06-30 20:39:45 +00:00
parent 40babd5582
commit cf8b27cd39
2 changed files with 18 additions and 14 deletions
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8
channels/chan_sip.c
View File

@@ -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 */
@@ -29308,12 +29307,17 @@ 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 */

2
configs/sip.conf.sample
View File

@@ -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