Merged revisions 325935 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines 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: http://svn.digium.com/svn/asterisk/trunk@325936 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -1335,7 +1335,6 @@ static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, con
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/*--- Misc functions */
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static void check_rtp_timeout(struct sip_pvt *dialog, time_t t);
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static int sip_do_reload(enum channelreloadreason reason);
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static int reload_config(enum channelreloadreason reason);
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static int expire_register(const void *data);
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static void *do_monitor(void *data);
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@ -13384,7 +13383,7 @@ static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xm
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return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
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}
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/*! \brief return the request and response heade for a 401 or 407 code */
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/*! \brief return the request and response header for a 401 or 407 code */
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static void auth_headers(enum sip_auth_type code, char **header, char **respheader)
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{
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if (code == WWW_AUTH) { /* 401 */
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@ -30003,6 +30002,7 @@ static int load_module(void)
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if (!(sip_tech.capabilities = ast_format_cap_alloc())) {
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return AST_MODULE_LOAD_FAILURE;
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}
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/* the fact that ao2_containers can't resize automatically is a major worry! */
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/* if the number of objects gets above MAX_XXX_BUCKETS, things will slow down */
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peers = ao2_t_container_alloc(HASH_PEER_SIZE, peer_hash_cb, peer_cmp_cb, "allocate peers");
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@ -30011,6 +30011,11 @@ static int load_module(void)
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dialogs_needdestroy = ao2_t_container_alloc(HASH_DIALOG_SIZE, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs_needdestroy");
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dialogs_rtpcheck = ao2_t_container_alloc(HASH_DIALOG_SIZE, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs for rtpchecks");
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threadt = ao2_t_container_alloc(HASH_DIALOG_SIZE, threadt_hash_cb, threadt_cmp_cb, "allocate threadt table");
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if (!peers || !peers_by_ip || !dialogs || !dialogs_needdestroy || !dialogs_rtpcheck
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|| !threadt) {
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ast_log(LOG_ERROR, "Unable to create primary SIP container(s)\n");
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return AST_MODULE_LOAD_FAILURE;
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}
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if (!(sip_cfg.caps = ast_format_cap_alloc())) {
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return AST_MODULE_LOAD_FAILURE;
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@ -130,7 +130,7 @@ allowoverlap=no ; Disable overlap dialing support. (Default is y
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; asterisk.conf, it defaults to that system name
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; Realms MUST be globally unique according to RFC 3261
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; Set this to your host name or domain name
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;domainsasrealm=no ; Use domans list as realms
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;domainsasrealm=no ; Use domains list as realms
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; You can serve multiple Realms specifying several
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; 'domain=...' directives (see below).
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; In this case Realm will be based on request 'From'/'To' header
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