[asterisk-commits] rmudgett: branch 1.8 r325935 - in /branches/1.8: channels/ configs/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Jun 30 15:39:49 CDT 2011
Author: rmudgett
Date: Thu Jun 30 15:39:45 2011
New Revision: 325935
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=325935
Log:
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.
Modified:
branches/1.8/channels/chan_sip.c
branches/1.8/configs/sip.conf.sample
Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=325935&r1=325934&r2=325935
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Thu Jun 30 15:39:45 2011
@@ -1319,7 +1319,6 @@
/*--- 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 @@
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 @@
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 @@
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 @@
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 @@
/* 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 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(®l); /* Registry object list -- not searched for anything */
ASTOBJ_CONTAINER_INIT(&submwil); /* MWI subscription object list */
@@ -29332,7 +29336,7 @@
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;
}
Modified: branches/1.8/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/configs/sip.conf.sample?view=diff&rev=325935&r1=325934&r2=325935
==============================================================================
--- branches/1.8/configs/sip.conf.sample (original)
+++ branches/1.8/configs/sip.conf.sample Thu Jun 30 15:39:45 2011
@@ -130,7 +130,7 @@
; 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
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