[svn-commits] oej: trunk r152019 - /trunk/channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sun Oct 26 04:45:18 CDT 2008
Author: oej
Date: Sun Oct 26 04:45:17 2008
New Revision: 152019
URL: http://svn.digium.com/view/asterisk?view=rev&rev=152019
Log:
Moving more variables to the sip_cfg structure, as I have some future ideas for the usage of that structure.
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=152019&r1=152018&r2=152019
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sun Oct 26 04:45:17 2008
@@ -766,6 +766,7 @@
#define DEFAULT_CALLERID "asterisk" /*!< Default caller ID */
#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
#define DEFAULT_ALLOWGUEST TRUE
+#define DEFAULT_RTPKEEPALIVE 0 /*!< Default RTPkeepalive setting */
#define DEFAULT_CALLCOUNTER FALSE
#define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */
#define DEFAULT_COMPACTHEADERS FALSE /*!< Send compact (one-character) SIP headers. Default off */
@@ -782,7 +783,10 @@
#define DEFAULT_NOTIFYRINGING TRUE /*!< Notify devicestate system on ringing state */
#define DEFAULT_PEDANTIC FALSE /*!< Avoid following SIP standards for dialog matching */
#define DEFAULT_AUTOCREATEPEER FALSE /*!< Don't create peers automagically */
+#define DEFAULT_MATCHEXTERNIPLOCALLY FALSE /*!< Match extern IP locally default setting */
#define DEFAULT_QUALIFY FALSE /*!< Don't monitor devices */
+#define DEFAULT_CALLEVENTS FALSE /*!< Extra manager SIP call events */
+#define DEFAULT_ALWAYSAUTHREJECT FALSE /*!< Don't reject authentication requests always */
#define DEFAULT_REGEXTENONQUALIFY FALSE
#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
#define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
@@ -815,14 +819,6 @@
static unsigned int default_transports; /*!< Default Transports (enum sip_transport) that are acceptable */
static unsigned int default_primary_transport; /*!< Default primary Transport (enum sip_transport) for outbound connections to devices */
-/*! \brief a place to store all global settings for the sip channel driver */
-struct sip_settings {
- int peer_rtupdate; /*!< G: Update database with registration data for peer? */
- int rtsave_sysname; /*!< G: Save system name at registration? */
- int ignore_regexpire; /*!< G: Ignore expiration of peer */
-};
-
-static struct sip_settings sip_cfg;
/*@}*/
/*! \name GlobalSettings
@@ -830,25 +826,42 @@
of sip.conf
*/
/*@{*/
-static int global_directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */
-static int global_rtautoclear; /*!< Realtime ?? */
+/*! \brief a place to store all global settings for the sip channel driver
+*/
+struct sip_settings {
+ int peer_rtupdate; /*!< G: Update database with registration data for peer? */
+ int rtsave_sysname; /*!< G: Save system name at registration? */
+ int ignore_regexpire; /*!< G: Ignore expiration of peer */
+ int rtautoclear; /*!< Realtime ?? */
+ int directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */
+ int pedanticsipchecking; /*!< Extra checking ? Default off */
+ int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
+ int srvlookup; /*!< SRV Lookup on or off. Default is on */
+ int allowguest; /*!< allow unauthenticated peers to connect? */
+ int alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
+ int compactheaders; /*!< send compact sip headers */
+ int allow_external_domains; /*!< Accept calls to external SIP domains? */
+ int callevents; /*!< Whether we send manager events or not */
+ int regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
+ int matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */
+};
+
+static struct sip_settings sip_cfg;
+
static int global_notifyringing; /*!< Send notifications on ringing */
static int global_notifyhold; /*!< Send notifications on hold */
-static int global_alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
-static int global_srvlookup; /*!< SRV Lookup on or off. Default is on */
-static int pedanticsipchecking; /*!< Extra checking ? Default off */
-static int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
+
static int global_relaxdtmf; /*!< Relax DTMF */
static int global_rtptimeout; /*!< Time out call if no RTP */
static int global_rtpholdtimeout; /*!< Time out call if no RTP during hold */
static int global_rtpkeepalive; /*!< Send RTP keepalives */
static int global_reg_timeout;
static int global_regattempts_max; /*!< Registration attempts before giving up */
-static int global_allowguest; /*!< allow unauthenticated peers to connect? */
static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer
call-limit to 999. When we remove the call-limit from the code, we can make it
with just a boolean flag in the device structure */
+static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */
static int global_allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
the global setting is in globals_flags[1] */
static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
@@ -859,7 +872,6 @@
static unsigned int global_cos_audio; /*!< 802.1p class of service for audio RTP packets */
static unsigned int global_cos_video; /*!< 802.1p class of service for video RTP packets */
static unsigned int global_cos_text; /*!< 802.1p class of service for text RTP packets */
-static int compactheaders; /*!< send compact sip headers */
static int recordhistory; /*!< Record SIP history. Off by default */
static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */
@@ -867,17 +879,12 @@
static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
static char global_sdpsession[AST_MAX_EXTENSION]; /*!< SDP session name for the SIP channel */
static char global_sdpowner[AST_MAX_EXTENSION]; /*!< SDP owner name for the SIP channel */
-static int allow_external_domains; /*!< Accept calls to external SIP domains? */
-static int global_callevents; /*!< Whether we send manager events or not */
static int global_authfailureevents; /*!< Whether we send authentication failure manager events or not. Default no. */
static int global_t1; /*!< T1 time */
static int global_t1min; /*!< T1 roundtrip time minimum */
static int global_timer_b; /*!< Timer B - RFC 3261 Section 17.1.1.2 */
-static int global_regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
static int global_autoframing; /*!< Turn autoframing on or off. */
-static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */
static struct sip_proxy global_outboundproxy; /*!< Outbound proxy */
-static int global_matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */
static int global_qualifyfreq; /*!< Qualify frequency */
@@ -2747,7 +2754,7 @@
if (!inet_aton(proxy->name, &proxy->ip.sin_addr)) {
/* Ok, not an IP address, then let's check if it's a domain or host */
/* XXX Todo - if we have proxy port, don't do SRV */
- if (ast_get_ip_or_srv(&proxy->ip, proxy->name, global_srvlookup ? "_sip._udp" : NULL) < 0) {
+ if (ast_get_ip_or_srv(&proxy->ip, proxy->name, sip_cfg.srvlookup ? "_sip._udp" : NULL) < 0) {
ast_log(LOG_WARNING, "Unable to locate host '%s'\n", proxy->name);
return FALSE;
}
@@ -3051,7 +3058,7 @@
ast_apply_ha(localaddr, &theirs) == AST_SENSE_ALLOW ;
if (want_remap &&
- (!global_matchexterniplocally || !ast_apply_ha(localaddr, us)) ) {
+ (!sip_cfg.matchexterniplocally || !ast_apply_ha(localaddr, us)) ) {
/* if we used externhost or stun, see if it is time to refresh the info */
if (externexpire && time(NULL) >= externexpire) {
if (stunaddr.sin_addr.s_addr) {
@@ -4211,7 +4218,7 @@
/* Cache peer */
ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
- AST_SCHED_REPLACE(peer->expire, sched, global_rtautoclear * 1000, expire_register, (void *) peer);
+ AST_SCHED_REPLACE(peer->expire, sched, sip_cfg.rtautoclear * 1000, expire_register, (void *) peer);
/* we could be incr. its refcount right here, but I guess, since
peers hang around until module unload time anyway, it's not worth the trouble */
}
@@ -4574,7 +4581,7 @@
*/
hostn = peername;
portno = port ? atoi(port) : (dialog->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
- if (global_srvlookup) {
+ if (sip_cfg.srvlookup) {
char service[MAXHOSTNAMELEN];
int tportno;
@@ -5902,7 +5909,7 @@
append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
/* Inform manager user about new channel and their SIP call ID */
- if (global_callevents)
+ if (sip_cfg.callevents)
manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
"Channel: %s\r\nUniqueid: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\n",
tmp->name, tmp->uniqueid, "SIP", i->callid, i->fullcontact);
@@ -6017,7 +6024,7 @@
for (x=*start; x<req->headers; x++) {
if (!strncasecmp(req->header[x], name, len)) {
char *r = req->header[x] + len; /* skip name */
- if (pedanticsipchecking)
+ if (sip_cfg.pedanticsipchecking)
r = ast_skip_blanks(r);
if (*r == ':') {
@@ -6384,7 +6391,7 @@
found = (!strcmp(p->callid, arg->callid));
else {
found = !strcmp(p->callid, arg->callid);
- if (pedanticsipchecking && found) {
+ if (sip_cfg.pedanticsipchecking && found) {
found = ast_strlen_zero(arg->tag) || ast_strlen_zero(p->theirtag) || !ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED) || !strcmp(p->theirtag, arg->tag);
}
}
@@ -6392,7 +6399,7 @@
ast_debug(5, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", p->callid, p->theirtag, p->tag);
/* If we get a new request within an existing to-tag - check the to tag as well */
- if (pedanticsipchecking && found && arg->method != SIP_RESPONSE) { /* SIP Request */
+ if (sip_cfg.pedanticsipchecking && found && arg->method != SIP_RESPONSE) { /* SIP Request */
if (p->tag[0] == '\0' && arg->totag[0]) {
/* We have no to tag, but they have. Wrong dialog */
found = FALSE;
@@ -6438,7 +6445,7 @@
arg.totag = totag;
arg.tag = ""; /* make sure tag is never NULL */
- if (pedanticsipchecking) {
+ if (sip_cfg.pedanticsipchecking) {
/* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy
we need more to identify a branch - so we have to check branch, from
and to tags to identify a call leg.
@@ -6466,7 +6473,7 @@
}
restartsearch:
- if (!pedanticsipchecking) {
+ if (!sip_cfg.pedanticsipchecking) {
struct sip_pvt tmp_dialog = {
.callid = callid,
};
@@ -7665,7 +7672,7 @@
ast_queue_frame(p->owner, &ast_null_frame);
/* Queue Manager Unhold event */
append_history(p, "Unhold", "%s", req->data->str);
- if (global_callevents)
+ if (sip_cfg.callevents)
manager_event(EVENT_FLAG_CALL, "Hold",
"Status: Off\r\n"
"Channel: %s\r\n"
@@ -7687,7 +7694,7 @@
ast_queue_frame(p->owner, &ast_null_frame);
/* Queue Manager Hold event */
append_history(p, "Hold", "%s", req->data->str);
- if (global_callevents && !ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
+ if (sip_cfg.callevents && !ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
manager_event(EVENT_FLAG_CALL, "Hold",
"Status: On\r\n"
"Channel: %s\r\n"
@@ -7732,7 +7739,7 @@
ast_str_append(&req->data, 0, "%s: %s\r\n", var, value);
req->header[req->headers] = req->data->str + req->len;
- if (compactheaders)
+ if (sip_cfg.compactheaders)
var = find_alias(var, var);
req->len += strlen(req->header[req->headers]);
req->headers++;
@@ -7915,7 +7922,7 @@
struct ast_hostent ahp;
int debug=sip_debug_test_pvt(p);
int tls_on = FALSE;
- int use_dns = global_srvlookup;
+ int use_dns = sip_cfg.srvlookup;
if (debug)
ast_verbose("set_destination: Parsing <%s> for address/port to send to\n", uri);
@@ -9367,7 +9374,7 @@
else /* Save for any further attempts */
ast_string_field_set(p, fromname, n);
- if (pedanticsipchecking) {
+ if (sip_cfg.pedanticsipchecking) {
ast_uri_encode(n, tmp_n, sizeof(tmp_n), 0);
n = tmp_n;
ast_uri_encode(l, tmp_l, sizeof(tmp_l), 0);
@@ -9389,7 +9396,7 @@
ast_str_append(&invite, 0, "sip:");
if (!ast_strlen_zero(p->username)) {
n = p->username;
- if (pedanticsipchecking) {
+ if (sip_cfg.pedanticsipchecking) {
ast_uri_encode(n, tmp_n, sizeof(tmp_n), 0);
n = tmp_n;
}
@@ -9615,7 +9622,7 @@
if (!mwi->dnsmgr) {
char transport[MAXHOSTNAMELEN];
snprintf(transport, sizeof(transport), "_sip._%s", get_transport(mwi->transport));
- ast_dnsmgr_lookup(mwi->hostname, &mwi->us, &mwi->dnsmgr, global_srvlookup ? transport : NULL);
+ ast_dnsmgr_lookup(mwi->hostname, &mwi->us, &mwi->dnsmgr, sip_cfg.srvlookup ? transport : NULL);
}
/* If we already have a subscription up simply send a resubscription */
@@ -10169,7 +10176,7 @@
if (r->dnsmgr == NULL) {
char transport[MAXHOSTNAMELEN];
snprintf(transport, sizeof(transport), "_sip._%s", get_transport(r->transport)); /* have to use static get_transport function */
- ast_dnsmgr_lookup(r->hostname, &r->us, &r->dnsmgr, global_srvlookup ? transport : NULL);
+ ast_dnsmgr_lookup(r->hostname, &r->us, &r->dnsmgr, sip_cfg.srvlookup ? transport : NULL);
}
if (r->call) { /* We have a registration */
@@ -11359,7 +11366,7 @@
terminate_uri(uri); /* warning, overwrite the string */
ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp));
- if (pedanticsipchecking)
+ if (sip_cfg.pedanticsipchecking)
ast_uri_decode(tmp);
c = get_in_brackets(tmp);
@@ -11459,7 +11466,7 @@
}
}
}
- if (!peer && autocreatepeer) {
+ if (!peer && sip_cfg.autocreatepeer) {
/* Create peer if we have autocreate mode enabled */
peer = temp_peer(name);
if (peer) {
@@ -11517,7 +11524,7 @@
case AUTH_NOT_FOUND:
case AUTH_PEER_NOT_DYNAMIC:
case AUTH_ACL_FAILED:
- if (global_alwaysauthreject) {
+ if (sip_cfg.alwaysauthreject) {
transmit_fake_auth_response(p, &p->initreq, 1);
} else {
/* URI not found */
@@ -11666,7 +11673,7 @@
if (req->rlPart2)
ast_copy_string(tmp, req->rlPart2, sizeof(tmp));
- if (pedanticsipchecking)
+ if (sip_cfg.pedanticsipchecking)
ast_uri_decode(tmp);
uri = get_in_brackets(tmp);
@@ -11686,7 +11693,7 @@
*/
ast_copy_string(tmpf, get_header(req, "From"), sizeof(tmpf));
if (!ast_strlen_zero(tmpf)) {
- if (pedanticsipchecking)
+ if (sip_cfg.pedanticsipchecking)
ast_uri_decode(tmpf);
from = get_in_brackets(tmpf);
}
@@ -11732,7 +11739,7 @@
domain_context[0] = '\0';
if (!check_sip_domain(p->domain, domain_context, sizeof(domain_context))) {
- if (!allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) {
+ if (!sip_cfg.allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) {
ast_debug(1, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain);
return -2;
}
@@ -11798,7 +11805,7 @@
if (sip_pvt_ptr) {
/* Go ahead and lock it (and its owner) before returning */
sip_pvt_lock(sip_pvt_ptr);
- if (pedanticsipchecking) {
+ if (sip_cfg.pedanticsipchecking) {
const char *pvt_fromtag, *pvt_totag;
unsigned char frommismatch = 0, tomismatch = 0;
@@ -11896,7 +11903,7 @@
}
h_refer_to = ast_strdupa(p_refer_to);
refer_to = get_in_brackets(h_refer_to);
- if (pedanticsipchecking)
+ if (sip_cfg.pedanticsipchecking)
ast_uri_decode(refer_to);
if (!strncasecmp(refer_to, "sip:", 4)) {
@@ -11923,7 +11930,7 @@
if (!ast_strlen_zero(p_referred_by)) {
char *lessthan;
h_referred_by = ast_strdupa(p_referred_by);
- if (pedanticsipchecking)
+ if (sip_cfg.pedanticsipchecking)
ast_uri_decode(h_referred_by);
/* Store referrer's caller ID name */
@@ -11982,7 +11989,7 @@
ast_copy_string(referdata->replaces_callid_fromtag, ptr, sizeof(referdata->replaces_callid_fromtag));
}
- if (!pedanticsipchecking)
+ if (!sip_cfg.pedanticsipchecking)
ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", referdata->replaces_callid );
else
ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", referdata->replaces_callid, referdata->replaces_callid_fromtag ? referdata->replaces_callid_fromtag : "<none>", referdata->replaces_callid_totag ? referdata->replaces_callid_totag : "<none>" );
@@ -12064,7 +12071,7 @@
ast_copy_string(tmp, get_header(req, "Also"), sizeof(tmp));
c = get_in_brackets(tmp);
- if (pedanticsipchecking)
+ if (sip_cfg.pedanticsipchecking)
ast_uri_decode(c);
if (!strncasecmp(c, "sip:", 4)) {
@@ -12471,7 +12478,7 @@
terminate_uri(uri2); /* trim extra stuff */
ast_copy_string(from, get_header(req, "From"), sizeof(from));
- if (pedanticsipchecking)
+ if (sip_cfg.pedanticsipchecking)
ast_uri_decode(from);
/* XXX here tries to map the username for invite things */
memset(calleridname, 0, sizeof(calleridname));
@@ -12562,10 +12569,10 @@
return res;
/* Finally, apply the guest policy */
- if (global_allowguest) {
+ if (sip_cfg.allowguest) {
replace_cid(p, rpid_num, calleridname);
res = AUTH_SUCCESSFUL;
- } else if (global_alwaysauthreject)
+ } else if (sip_cfg.alwaysauthreject)
res = AUTH_FAKE_AUTH; /* reject with fake authorization request */
else
res = AUTH_SECRET_FAILED; /* we don't want any guests, authentication will fail */
@@ -14126,9 +14133,9 @@
}
ast_cli(a->fd, " Videosupport: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT)));
ast_cli(a->fd, " Textsupport: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT)));
- ast_cli(a->fd, " AutoCreate Peer: %s\n", cli_yesno(autocreatepeer));
+ ast_cli(a->fd, " AutoCreate Peer: %s\n", cli_yesno(sip_cfg.autocreatepeer));
ast_cli(a->fd, " Match Auth Username: %s\n", cli_yesno(global_match_auth_username));
- ast_cli(a->fd, " Allow unknown access: %s\n", cli_yesno(global_allowguest));
+ ast_cli(a->fd, " Allow unknown access: %s\n", cli_yesno(sip_cfg.allowguest));
ast_cli(a->fd, " Allow subscriptions: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
ast_cli(a->fd, " Allow overlap dialing: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)));
ast_cli(a->fd, " Allow promsic. redir: %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_PROMISCREDIR)));
@@ -14136,19 +14143,19 @@
ast_cli(a->fd, " SIP domain support: %s\n", cli_yesno(!AST_LIST_EMPTY(&domain_list)));
ast_cli(a->fd, " Realm. auth: %s\n", cli_yesno(authl != NULL));
ast_cli(a->fd, " Our auth realm %s\n", global_realm);
- ast_cli(a->fd, " Call to non-local dom.: %s\n", cli_yesno(allow_external_domains));
+ ast_cli(a->fd, " Call to non-local dom.: %s\n", cli_yesno(sip_cfg.allow_external_domains));
ast_cli(a->fd, " URI user is phone no: %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_USEREQPHONE)));
- ast_cli(a->fd, " Always auth rejects: %s\n", cli_yesno(global_alwaysauthreject));
- ast_cli(a->fd, " Direct RTP setup: %s\n", cli_yesno(global_directrtpsetup));
+ ast_cli(a->fd, " Always auth rejects: %s\n", cli_yesno(sip_cfg.alwaysauthreject));
+ ast_cli(a->fd, " Direct RTP setup: %s\n", cli_yesno(sip_cfg.directrtpsetup));
ast_cli(a->fd, " User Agent: %s\n", global_useragent);
ast_cli(a->fd, " SDP Session Name: %s\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
ast_cli(a->fd, " SDP Owner Name: %s\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner);
ast_cli(a->fd, " Reg. context: %s\n", S_OR(global_regcontext, "(not set)"));
- ast_cli(a->fd, " Regexten on Qualify: %s\n", cli_yesno(global_regextenonqualify));
+ ast_cli(a->fd, " Regexten on Qualify: %s\n", cli_yesno(sip_cfg.regextenonqualify));
ast_cli(a->fd, " Caller ID: %s\n", default_callerid);
ast_cli(a->fd, " From: Domain: %s\n", default_fromdomain);
ast_cli(a->fd, " Record SIP history: %s\n", recordhistory ? "On" : "Off");
- ast_cli(a->fd, " Call Events: %s\n", global_callevents ? "On" : "Off");
+ ast_cli(a->fd, " Call Events: %s\n", sip_cfg.callevents ? "On" : "Off");
ast_cli(a->fd, " Auth. Failure Events: %s\n", global_authfailureevents ? "On" : "Off");
ast_cli(a->fd, " T38 fax pt UDPTL: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL)));
@@ -14219,13 +14226,13 @@
ast_cli(a->fd, "\n");
ast_cli(a->fd, " Relax DTMF: %s\n", cli_yesno(global_relaxdtmf));
ast_cli(a->fd, " RFC2833 Compensation: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_RFC2833_COMPENSATE)));
- ast_cli(a->fd, " Compact SIP headers: %s\n", cli_yesno(compactheaders));
+ ast_cli(a->fd, " Compact SIP headers: %s\n", cli_yesno(sip_cfg.compactheaders));
ast_cli(a->fd, " RTP Keepalive: %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" );
ast_cli(a->fd, " RTP Timeout: %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" );
ast_cli(a->fd, " RTP Hold Timeout: %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)");
ast_cli(a->fd, " MWI NOTIFY mime type: %s\n", default_notifymime);
- ast_cli(a->fd, " DNS SRV lookup: %s\n", cli_yesno(global_srvlookup));
- ast_cli(a->fd, " Pedantic SIP support: %s\n", cli_yesno(pedanticsipchecking));
+ ast_cli(a->fd, " DNS SRV lookup: %s\n", cli_yesno(sip_cfg.srvlookup));
+ ast_cli(a->fd, " Pedantic SIP support: %s\n", cli_yesno(sip_cfg.pedanticsipchecking));
ast_cli(a->fd, " Reg. min duration %d secs\n", min_expiry);
ast_cli(a->fd, " Reg. max duration: %d secs\n", max_expiry);
ast_cli(a->fd, " Reg. default duration: %d secs\n", default_expiry);
@@ -14271,7 +14278,7 @@
ast_cli(a->fd, " Update: %s\n", cli_yesno(sip_cfg.peer_rtupdate));
ast_cli(a->fd, " Ignore Reg. Expire: %s\n", cli_yesno(sip_cfg.ignore_regexpire));
ast_cli(a->fd, " Save sys. name: %s\n", cli_yesno(sip_cfg.rtsave_sysname));
- ast_cli(a->fd, " Auto Clear: %d\n", global_rtautoclear);
+ ast_cli(a->fd, " Auto Clear: %d\n", sip_cfg.rtautoclear);
}
ast_cli(a->fd, "\n----\n");
return CLI_SUCCESS;
@@ -15955,7 +15962,7 @@
if (!req->ignore && p->owner) {
if (!reinvite) {
ast_queue_control(p->owner, AST_CONTROL_ANSWER);
- if (global_callevents)
+ if (sip_cfg.callevents)
manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
"Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
p->owner->name, p->owner->uniqueid, "SIP", p->callid, p->fullcontact, p->peername);
@@ -16494,7 +16501,7 @@
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
"ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: %s\r\nTime: %d\r\n",
peer->name, s, pingtime);
- if (is_reachable && global_regextenonqualify)
+ if (is_reachable && sip_cfg.regextenonqualify)
register_peer_exten(peer, TRUE);
}
@@ -17917,7 +17924,7 @@
/* If pedantic is on, we need to check the tags. If they're different, this is
in fact a forked call through a SIP proxy somewhere. */
int different;
- if (pedanticsipchecking)
+ if (sip_cfg.pedanticsipchecking)
different = sip_uri_cmp(p->initreq.rlPart2, req->rlPart2);
else
different = strcmp(p->initreq.rlPart2, req->rlPart2);
@@ -18825,7 +18832,7 @@
ast_string_field_set(p, context, default_context);
/* If we do not support SIP domains, all transfers are local */
- if (allow_external_domains && check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
+ if (sip_cfg.allow_external_domains && check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
p->refer->localtransfer = 1;
if (sipdebug)
ast_debug(3, "This SIP transfer is local : %s\n", p->refer->refer_to_domain);
@@ -19811,7 +19818,7 @@
}
snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd);
- if (pedanticsipchecking) {
+ if (sip_cfg.pedanticsipchecking) {
/* If this is a request packet without a from tag, it's not
correct according to RFC 3261 */
/* Check if this a new request in a new dialog with a totag already attached to it,
@@ -19967,7 +19974,7 @@
if (sip_debug_test_addr(sin)) /* Set the debug flag early on packet level */
req->debug = 1;
- if (pedanticsipchecking)
+ if (sip_cfg.pedanticsipchecking)
req->len = lws2sws(req->data->str, req->len); /* Fix multiline headers */
if (req->debug) {
ast_verbose("\n<--- SIP read from %s:%s:%d --->\n%s\n<------------->\n",
@@ -20774,7 +20781,7 @@
if (peer->lastms > -1) {
ast_log(LOG_NOTICE, "Peer '%s' is now UNREACHABLE! Last qualify: %d\n", peer->name, peer->lastms);
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unreachable\r\nTime: %d\r\n", peer->name, -1);
- if (global_regextenonqualify)
+ if (sip_cfg.regextenonqualify)
register_peer_exten(peer, FALSE);
}
if (peer->call) {
@@ -21129,7 +21136,7 @@
p->jointcapability = oldformat;
sip_pvt_lock(p);
tmpc = sip_new(p, AST_STATE_DOWN, host); /* Place the call */
- if (global_callevents)
+ if (sip_cfg.callevents)
manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
"Channel: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
p->owner? p->owner->name : "", "SIP", p->callid, p->fullcontact, p->peername);
@@ -21953,7 +21960,7 @@
snprintf(transport, sizeof(transport), "_sip._%s", get_transport(peer->socket.type));
- if (ast_dnsmgr_lookup(_srvlookup, &peer->addr, &peer->dnsmgr, global_srvlookup ? transport : NULL)) {
+ if (ast_dnsmgr_lookup(_srvlookup, &peer->addr, &peer->dnsmgr, sip_cfg.srvlookup ? transport : NULL)) {
unref_peer(peer, "getting rid of a peer pointer");
return NULL;
}
@@ -22158,7 +22165,7 @@
ourport_tcp = STANDARD_SIP_PORT;
ourport_tls = STANDARD_TLS_PORT;
bindaddr.sin_port = htons(STANDARD_SIP_PORT);
- global_srvlookup = DEFAULT_SRVLOOKUP;
+ sip_cfg.srvlookup = DEFAULT_SRVLOOKUP;
global_tos_sip = DEFAULT_TOS_SIP;
global_tos_audio = DEFAULT_TOS_AUDIO;
global_tos_video = DEFAULT_TOS_VIDEO;
@@ -22173,13 +22180,13 @@
externrefresh = 10;
/* Reset channel settings to default before re-configuring */
- allow_external_domains = DEFAULT_ALLOW_EXT_DOM; /* Allow external invites */
+ sip_cfg.allow_external_domains = DEFAULT_ALLOW_EXT_DOM; /* Allow external invites */
global_regcontext[0] = '\0';
- global_regextenonqualify = DEFAULT_REGEXTENONQUALIFY;
+ sip_cfg.regextenonqualify = DEFAULT_REGEXTENONQUALIFY;
global_notifyringing = DEFAULT_NOTIFYRINGING;
global_notifyhold = FALSE; /*!< Keep track of hold status for a peer */
- global_directrtpsetup = FALSE; /* Experimental feature, disabled by default */
- global_alwaysauthreject = 0;
+ sip_cfg.directrtpsetup = FALSE; /* Experimental feature, disabled by default */
+ sip_cfg.alwaysauthreject = DEFAULT_ALWAYSAUTHREJECT;
global_allowsubscribe = FALSE;
snprintf(global_useragent, sizeof(global_useragent), "%s %s", DEFAULT_USERAGENT, ast_get_version());
snprintf(global_sdpsession, sizeof(global_sdpsession), "%s %s", DEFAULT_SDPSESSION, ast_get_version());
@@ -22187,20 +22194,20 @@
ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime));
ast_copy_string(global_realm, S_OR(ast_config_AST_SYSTEM_NAME, DEFAULT_REALM), sizeof(global_realm));
ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid));
- compactheaders = DEFAULT_COMPACTHEADERS;
+ sip_cfg.compactheaders = DEFAULT_COMPACTHEADERS;
global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
global_regattempts_max = 0;
- pedanticsipchecking = DEFAULT_PEDANTIC;
- autocreatepeer = DEFAULT_AUTOCREATEPEER;
+ sip_cfg.pedanticsipchecking = DEFAULT_PEDANTIC;
+ sip_cfg.autocreatepeer = DEFAULT_AUTOCREATEPEER;
global_autoframing = 0;
- global_allowguest = DEFAULT_ALLOWGUEST;
+ sip_cfg.allowguest = DEFAULT_ALLOWGUEST;
global_callcounter = DEFAULT_CALLCOUNTER;
global_match_auth_username = FALSE; /*!< Match auth username if available instead of From: Default off. */
global_rtptimeout = 0;
global_rtpholdtimeout = 0;
- global_rtpkeepalive = 0;
+ global_rtpkeepalive = DEFAULT_RTPKEEPALIVE;
global_allowtransfer = TRANSFER_OPENFORALL; /* Merrily accept all transfers by default */
- global_rtautoclear = 120;
+ sip_cfg.rtautoclear = 120;
ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE); /* Default for all devices: TRUE */
ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP); /* Default for all devices: TRUE */
sip_cfg.peer_rtupdate = TRUE;
@@ -22232,14 +22239,14 @@
/* Misc settings for the channel */
global_relaxdtmf = FALSE;
- global_callevents = FALSE;
+ sip_cfg.callevents = DEFAULT_CALLEVENTS;
global_authfailureevents = FALSE;
global_t1 = SIP_TIMER_T1;
global_timer_b = 64 * SIP_TIMER_T1;
global_t1min = DEFAULT_T1MIN;
global_qualifyfreq = DEFAULT_QUALIFYFREQ;
- global_matchexterniplocally = FALSE;
+ sip_cfg.matchexterniplocally = DEFAULT_MATCHEXTERNIPLOCALLY;
/* Copy the default jb config over global_jbconf */
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
@@ -22263,7 +22270,7 @@
} else if (!strcasecmp(v->name, "callcounter")) {
global_callcounter = ast_true(v->value) ? 1 : 0;
} else if (!strcasecmp(v->name, "allowguest")) {
- global_allowguest = ast_true(v->value) ? 1 : 0;
+ sip_cfg.allowguest = ast_true(v->value) ? 1 : 0;
} else if (!strcasecmp(v->name, "realm")) {
ast_copy_string(global_realm, v->value, sizeof(global_realm));
} else if (!strcasecmp(v->name, "useragent")) {
@@ -22357,7 +22364,7 @@
} else if (!strcasecmp(v->name, "rtautoclear")) {
int i = atoi(v->value);
if (i > 0)
- global_rtautoclear = i;
+ sip_cfg.rtautoclear = i;
else
i = 0;
ast_set2_flag(&global_flags[1], i || ast_true(v->value), SIP_PAGE2_RTAUTOCLEAR);
@@ -22380,20 +22387,20 @@
} else if (!strcasecmp(v->name, "rtpkeepalive")) {
if ((sscanf(v->value, "%d", &global_rtpkeepalive) != 1) || (global_rtpkeepalive < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d. Using default.\n", v->value, v->lineno);
- global_rtpkeepalive = 0;
+ global_rtpkeepalive = DEFAULT_RTPKEEPALIVE;
}
} else if (!strcasecmp(v->name, "compactheaders")) {
- compactheaders = ast_true(v->value);
+ sip_cfg.compactheaders = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifymimetype")) {
ast_copy_string(default_notifymime, v->value, sizeof(default_notifymime));
} else if (!strcasecmp(v->name, "directrtpsetup")) {
- global_directrtpsetup = ast_true(v->value);
+ sip_cfg.directrtpsetup = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifyringing")) {
global_notifyringing = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifyhold")) {
global_notifyhold = ast_true(v->value);
} else if (!strcasecmp(v->name, "alwaysauthreject")) {
- global_alwaysauthreject = ast_true(v->value);
+ sip_cfg.alwaysauthreject = ast_true(v->value);
} else if (!strcasecmp(v->name, "mohinterpret")) {
ast_copy_string(default_mohinterpret, v->value, sizeof(default_mohinterpret));
} else if (!strcasecmp(v->name, "mohsuggest")) {
@@ -22412,7 +22419,7 @@
}
ast_copy_string(global_regcontext, v->value, sizeof(global_regcontext));
} else if (!strcasecmp(v->name, "regextenonqualify")) {
- global_regextenonqualify = ast_true(v->value);
+ sip_cfg.regextenonqualify = ast_true(v->value);
} else if (!strcasecmp(v->name, "callerid")) {
ast_copy_string(default_callerid, v->value, sizeof(default_callerid));
} else if (!strcasecmp(v->name, "fromdomain")) {
@@ -22446,13 +22453,13 @@
ast_copy_string(global_outboundproxy.name, proxyname, sizeof(global_outboundproxy.name));
} else if (!strcasecmp(v->name, "autocreatepeer")) {
- autocreatepeer = ast_true(v->value);
+ sip_cfg.autocreatepeer = ast_true(v->value);
} else if (!strcasecmp(v->name, "match_auth_username")) {
global_match_auth_username = ast_true(v->value);
} else if (!strcasecmp(v->name, "srvlookup")) {
- global_srvlookup = ast_true(v->value);
+ sip_cfg.srvlookup = ast_true(v->value);
} else if (!strcasecmp(v->name, "pedantic")) {
- pedanticsipchecking = ast_true(v->value);
+ sip_cfg.pedanticsipchecking = ast_true(v->value);
} else if (!strcasecmp(v->name, "maxexpirey") || !strcasecmp(v->name, "maxexpiry")) {
max_expiry = atoi(v->value);
if (max_expiry < 1)
@@ -22528,7 +22535,7 @@
} else if (!strcasecmp(v->name, "autoframing")) {
global_autoframing = ast_true(v->value);
} else if (!strcasecmp(v->name, "allowexternaldomains")) {
- allow_external_domains = ast_true(v->value);
+ sip_cfg.allow_external_domains = ast_true(v->value);
} else if (!strcasecmp(v->name, "autodomain")) {
auto_sip_domains = ast_true(v->value);
} else if (!strcasecmp(v->name, "domain")) {
@@ -22619,7 +22626,7 @@
global_qualifyfreq = DEFAULT_QUALIFYFREQ;
}
} else if (!strcasecmp(v->name, "callevents")) {
- global_callevents = ast_true(v->value);
+ sip_cfg.callevents = ast_true(v->value);
} else if (!strcasecmp(v->name, "authfailureevents")) {
global_authfailureevents = ast_true(v->value);
} else if (!strcasecmp(v->name, "maxcallbitrate")) {
@@ -22627,7 +22634,7 @@
if (default_maxcallbitrate < 0)
default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
} else if (!strcasecmp(v->name, "matchexterniplocally")) {
- global_matchexterniplocally = ast_true(v->value);
+ sip_cfg.matchexterniplocally = ast_true(v->value);
} else if (!strcasecmp(v->name, "session-timers")) {
int i = (int) str2stmode(v->value);
if (i < 0) {
@@ -22661,9 +22668,9 @@
}
}
- if (!allow_external_domains && AST_LIST_EMPTY(&domain_list)) {
+ if (!sip_cfg.allow_external_domains && AST_LIST_EMPTY(&domain_list)) {
ast_log(LOG_WARNING, "To disallow external domains, you need to configure local SIP domains.\n");
- allow_external_domains = 1;
+ sip_cfg.allow_external_domains = 1;
}
/* If not configured, set default transports */
if (default_transports == 0) {
@@ -23110,7 +23117,7 @@
return -1;
/* Disable early RTP bridge */
- if (chan->_state != AST_STATE_UP && !global_directrtpsetup) /* We are in early state */
+ if (chan->_state != AST_STATE_UP && !sip_cfg.directrtpsetup) /* We are in early state */
return 0;
sip_pvt_lock(p);
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