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Merged revisions 325935 via svnmerge from

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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: https://origsvn.digium.com/svn/asterisk/trunk@325936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
10-digiumphones
Richard Mudgett 15 years ago
parent 1339a0a535
commit 39a7152df3
2 changed files with 18 additions and 13 deletions
Show Stats Download Patch File Download Diff File

29
channels/chan_sip.c
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@ -1335,7 +1335,6 @@ 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);
@ -13384,7 +13383,7 @@ 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 */
@ -17357,13 +17356,13 @@ 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));
@ -17457,13 +17456,13 @@ 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\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)) ? "Y" : "N");
@ -19093,7 +19092,7 @@ 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 */
@ -19117,7 +19116,7 @@ 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 */
@ -30003,6 +30002,7 @@ static int load_module(void)
if (!(sip_tech.capabilities = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_FAILURE;
}
/* 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");
@ -30011,6 +30011,11 @@ static int load_module(void)
dialogs_needdestroy = ao2_t_container_alloc(HASH_DIALOG_SIZE, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs_needdestroy");
dialogs_rtpcheck = ao2_t_container_alloc(HASH_DIALOG_SIZE, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs for rtpchecks");
threadt = ao2_t_container_alloc(HASH_DIALOG_SIZE, threadt_hash_cb, threadt_cmp_cb, "allocate threadt table");
if (!peers || !peers_by_ip || !dialogs || !dialogs_needdestroy || !dialogs_rtpcheck
|| !threadt) {
ast_log(LOG_ERROR, "Unable to create primary SIP container(s)\n");
return AST_MODULE_LOAD_FAILURE;
}
if (!(sip_cfg.caps = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_FAILURE;
@ -30034,7 +30039,7 @@ 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;
}

2
configs/sip.conf.sample
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@ -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

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