[asterisk-commits] trunk - r8514 /trunk/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Jan 24 03:19:14 MST 2006
Author: oej
Date: Tue Jan 24 04:19:13 2006
New Revision: 8514
URL: http://svn.digium.com/view/asterisk?rev=8514&view=rev
Log:
- Adding doxygen comments
- Changing default values set in reload_config to DEFAULT_ #defines to make it more clear what defaults are
- Cleaning up global_ and default_ variable naming.
- Moving variable and #defines together in the source, adding comments to explain sections
Global_ is used for channel settings that does not apply to peers or users as defaults for their settings
default_ is used both as a channel setting for unknown callers, as well as defaults for peers and users
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=8514&r1=8513&r2=8514&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Jan 24 04:19:13 2006
@@ -92,8 +92,12 @@
#include "asterisk/astosp.h"
#endif
-#ifndef DEFAULT_USERAGENT
-#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
+#ifndef FALSE
+#define FALSE 0
+#endif
+
+#ifndef TRUE
+#define TRUE 1
#endif
#define VIDEO_CODEC_MASK 0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
@@ -111,19 +115,21 @@
/* guard limit must be larger than guard secs */
/* guard min must be < 1000, and should be >= 250 */
-#define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
-#define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of
- EXPIRY_GUARD_SECS */
-#define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
- GUARD_PCT turns out to be lower than this, it
- will use this time instead.
- This is in milliseconds. */
-#define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
- below EXPIRY_GUARD_LIMIT */
+#define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
+#define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of
+ EXPIRY_GUARD_SECS */
+#define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
+ GUARD_PCT turns out to be lower than this, it
+ will use this time instead.
+ This is in milliseconds. */
+#define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
+ below EXPIRY_GUARD_LIMIT */
+#define DEFAULT_EXPIRY 900 /*!< Expire slowly */
static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
+static int expiry = DEFAULT_EXPIRY;
#ifndef MAX
#define MAX(a,b) ((a) > (b) ? (a) : (b))
@@ -133,13 +139,16 @@
-#define DEFAULT_MAXMS 2000 /*!< Must be faster than 2 seconds by default */
-#define DEFAULT_FREQ_OK 60 * 1000 /*!< How often to check for the host to be up */
-#define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< How often to check, if the host is down... */
+#define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
+#define DEFAULT_FREQ_OK 60 * 1000 /*!< Qualification: How often to check for the host to be up */
+#define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< Qualification: How often to check, if the host is down... */
#define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
#define MAX_RETRANS 6 /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
#define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
+
+#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
+#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
static const char desc[] = "Session Initiation Protocol (SIP)";
@@ -326,76 +335,90 @@
/*! \brief SIP Extensions we support */
#define SUPPORTED_EXTENSIONS "replaces"
+#define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */
+
+/* Default values, set and reset in reload_config before reading configuration */
+/* These are default values in the source. There are other recommended values in the
+ sip.conf.sample for new installations. These may differ to keep backwards compatibility,
+ yet encouraging new behaviour on new installations
+ */
#define DEFAULT_SIP_PORT 5060 /*!< From RFC 3261 (former 2543) */
-#define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */
-
-static char default_useragent[AST_MAX_EXTENSION] = DEFAULT_USERAGENT;
-
-#define DEFAULT_CONTEXT "default"
+#define DEFAULT_CONTEXT "default"
+#define DEFAULT_MUSICCLASS "default"
+#define DEFAULT_VMEXTEN "asterisk"
+#define DEFAULT_CALLERID "asterisk"
+#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
+#define DEFAULT_MWITIME 10
+#define DEFAULT_ALLOWGUEST TRUE
+#define DEFAULT_VIDEOSUPPORT FALSE
+#define DEFAULT_SRVLOOKUP FALSE /*!< Recommended setting is ON */
+#define DEFAULT_COMPACTHEADERS FALSE
+#define DEFAULT_TOS FALSE
+#define DEFAULT_ALLOW_EXT_DOM TRUE
+#define DEFAULT_REALM "asterisk"
+#define DEFAULT_NOTIFYRINGING TRUE
+#define DEFAULT_PEDANTIC FALSE
+#define DEFAULT_AUTOCREATEPEER FALSE
+#define DEFAULT_QUALIFY FALSE
+#ifndef DEFAULT_USERAGENT
+#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
+#endif
+
+/* Default setttings are used as a channel setting and as a default when
+ configuring devices */
static char default_context[AST_MAX_CONTEXT];
static char default_subscribecontext[AST_MAX_CONTEXT];
-
-#define DEFAULT_VMEXTEN "asterisk"
-static char global_vmexten[AST_MAX_EXTENSION];
-
static char default_language[MAX_LANGUAGE];
-
-#define DEFAULT_CALLERID "asterisk"
static char default_callerid[AST_MAX_EXTENSION];
-
static char default_fromdomain[AST_MAX_EXTENSION];
-
-#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
static char default_notifymime[AST_MAX_EXTENSION];
-
+static int default_qualify; /*!< Default Qualify= setting */
+static char default_vmexten[AST_MAX_EXTENSION];
+static char default_musicclass[MAX_MUSICCLASS]; /*!< Global music on hold class */
+
+/* Global settings only apply to the channel */
static int global_notifyringing; /*!< Send notifications on ringing */
-
-static int default_qualify; /*!< Default Qualify= setting */
-
-
static int srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
-
static int pedanticsipchecking; /*!< Extra checking ? Default off */
-
static int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
-
static int relaxdtmf; /*!< Relax DTMF */
-
static int global_rtptimeout; /*!< Time out call if no RTP */
-
static int global_rtpholdtimeout;
-
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 = 1; /*!< allow unauthenticated users/peers to connect? */
-
-#define DEFAULT_MWITIME 10
+static int global_allowguest; /*!< allow unauthenticated users/peers to connect? */
static int global_mwitime; /*!< Time between MWI checks for peers */
-
-static int tos = 0;
-
-static int videosupport = 0;
-
-static int compactheaders = 0; /*!< send compact sip headers */
-
+static int global_tos; /*!< IP Type of service */
+static int global_videosupport; /*!< Videosupport on or off */
+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 */
+static char regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
+static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
+static int allow_external_domains; /*!< Accept calls to external SIP domains? */
+
+/*! \brief Codecs that we support by default: */
+static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
+static int noncodeccapability = AST_RTP_DTMF;
/* Object counters */
-static int suserobjs = 0;
-static int ruserobjs = 0;
-static int speerobjs = 0;
-static int rpeerobjs = 0;
-static int apeerobjs = 0;
-static int regobjs = 0;
+static int suserobjs = 0; /*!< Static users */
+static int ruserobjs = 0; /*!< Realtime users */
+static int speerobjs = 0; /*!< Statis peers */
+static int rpeerobjs = 0; /*!< Realtime peers */
+static int apeerobjs = 0; /*!< Autocreated peer objects */
+static int regobjs = 0; /*!< Registry objects */
static struct ast_flags global_flags = {0}; /*!< global SIP_ flags */
static struct ast_flags global_flags_page2 = {0}; /*!< more global SIP_ flags */
static int usecnt =0;
+
AST_MUTEX_DEFINE_STATIC(usecnt_lock);
-AST_MUTEX_DEFINE_STATIC(rand_lock);
+AST_MUTEX_DEFINE_STATIC(rand_lock); /*!< Lock for thread-safe random generator */
/*! \brief Protect the SIP dialog list (of sip_pvt's) */
AST_MUTEX_DEFINE_STATIC(iflock);
@@ -412,32 +435,16 @@
static int restart_monitor(void);
-/*! \brief Codecs that we support by default: */
-static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
-static int noncodeccapability = AST_RTP_DTMF;
static struct in_addr __ourip;
static struct sockaddr_in outboundproxyip;
static int ourport;
-
static struct sockaddr_in debugaddr;
-static int recordhistory; /*!< Record SIP history. Off by default */
-static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
-
-static char global_musicclass[MAX_MUSICCLASS]; /*!< Global music on hold class */
-#define DEFAULT_REALM "asterisk"
-static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */
-static char regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
-
-#define DEFAULT_EXPIRY 900 /*!< Expire slowly */
-static int expiry = DEFAULT_EXPIRY;
static struct sched_context *sched;
static struct io_context *io;
-#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
-#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
#define DEC_CALL_LIMIT 0
#define INC_CALL_LIMIT 1
@@ -493,7 +500,6 @@
static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */
-int allow_external_domains; /*!< Accept calls to external SIP domains? */
/*! \brief sip_history: Structure for saving transactions within a SIP dialog */
struct sip_history {
@@ -3129,10 +3135,10 @@
if (sip_methods[intended_method].need_rtp) {
p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
- if (videosupport)
+ if (global_videosupport)
p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
- if (!p->rtp || (videosupport && !p->vrtp)) {
- ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n", videosupport ? "and video" : "", strerror(errno));
+ if (!p->rtp || (global_videosupport && !p->vrtp)) {
+ ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n", global_videosupport ? "and video" : "", strerror(errno));
ast_mutex_destroy(&p->lock);
if (p->chanvars) {
ast_variables_destroy(p->chanvars);
@@ -3141,9 +3147,9 @@
free(p);
return NULL;
}
- ast_rtp_settos(p->rtp, tos);
+ ast_rtp_settos(p->rtp, global_tos);
if (p->vrtp)
- ast_rtp_settos(p->vrtp, tos);
+ ast_rtp_settos(p->vrtp, global_tos);
p->rtptimeout = global_rtptimeout;
p->rtpholdtimeout = global_rtpholdtimeout;
p->rtpkeepalive = global_rtpkeepalive;
@@ -3168,7 +3174,7 @@
ast_string_field_set(p, callid, callid);
ast_copy_flags(p, &global_flags, SIP_FLAGS_TO_COPY);
/* Assign default music on hold class */
- ast_string_field_set(p, musicclass, global_musicclass);
+ ast_string_field_set(p, musicclass, default_musicclass);
p->capability = global_capability;
if ((ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833) || (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_AUTO))
p->noncodeccapability |= AST_RTP_DTMF;
@@ -4071,7 +4077,7 @@
add_header(resp, "To", ot);
copy_header(resp, req, "Call-ID");
copy_header(resp, req, "CSeq");
- add_header(resp, "User-Agent", default_useragent);
+ add_header(resp, "User-Agent", global_useragent);
add_header(resp, "Allow", ALLOWED_METHODS);
if (msg[0] == '2' && (p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER)) {
/* For registration responses, we also need expiry and
@@ -4187,7 +4193,7 @@
copy_header(req, orig, "Call-ID");
add_header(req, "CSeq", tmp);
- add_header(req, "User-Agent", default_useragent);
+ add_header(req, "User-Agent", global_useragent);
add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
if (!ast_strlen_zero(p->rpid))
@@ -4506,7 +4512,7 @@
}
/* Now send any other common codecs, and non-codec formats: */
- for (x = 1; x <= ((videosupport && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) {
+ for (x = 1; x <= ((global_videosupport && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) {
if (!(capability & x))
continue;
@@ -4916,7 +4922,7 @@
add_header(req, "Contact", p->our_contact);
add_header(req, "Call-ID", p->callid);
add_header(req, "CSeq", tmp);
- add_header(req, "User-Agent", default_useragent);
+ add_header(req, "User-Agent", global_useragent);
add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
if (!ast_strlen_zero(p->rpid))
add_header(req, "Remote-Party-ID", p->rpid);
@@ -5205,7 +5211,7 @@
add_header(&req, "Content-Type", default_notifymime);
ast_build_string(&t, &maxbytes, "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no");
- ast_build_string(&t, &maxbytes, "Message-Account: sip:%s@%s\r\n", !ast_strlen_zero(vmexten) ? vmexten : global_vmexten, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain);
+ ast_build_string(&t, &maxbytes, "Message-Account: sip:%s@%s\r\n", !ast_strlen_zero(vmexten) ? vmexten : default_vmexten, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain);
ast_build_string(&t, &maxbytes, "Voice-Message: %d/%d (0/0)\r\n", newmsgs, oldmsgs);
if (t > tmp + sizeof(tmp))
@@ -5511,7 +5517,7 @@
add_header(&req, "To", to);
add_header(&req, "Call-ID", p->callid);
add_header(&req, "CSeq", tmp);
- add_header(&req, "User-Agent", default_useragent);
+ add_header(&req, "User-Agent", global_useragent);
add_header(&req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
@@ -8247,7 +8253,7 @@
ast_cli(fd, "----------------\n");
ast_cli(fd, " SIP Port: %d\n", ntohs(bindaddr.sin_port));
ast_cli(fd, " Bindaddress: %s\n", ast_inet_ntoa(tmp, sizeof(tmp), bindaddr.sin_addr));
- ast_cli(fd, " Videosupport: %s\n", videosupport ? "Yes" : "No");
+ ast_cli(fd, " Videosupport: %s\n", global_videosupport ? "Yes" : "No");
ast_cli(fd, " AutoCreatePeer: %s\n", autocreatepeer ? "Yes" : "No");
ast_cli(fd, " Allow unknown access: %s\n", global_allowguest ? "Yes" : "No");
ast_cli(fd, " Promsic. redir: %s\n", ast_test_flag(&global_flags, SIP_PROMISCREDIR) ? "Yes" : "No");
@@ -8256,14 +8262,14 @@
ast_cli(fd, " URI user is phone no: %s\n", ast_test_flag(&global_flags, SIP_USEREQPHONE) ? "Yes" : "No");
ast_cli(fd, " Our auth realm %s\n", global_realm);
ast_cli(fd, " Realm. auth: %s\n", authl ? "Yes": "No");
- ast_cli(fd, " User Agent: %s\n", default_useragent);
+ ast_cli(fd, " User Agent: %s\n", global_useragent);
ast_cli(fd, " MWI checking interval: %d secs\n", global_mwitime);
ast_cli(fd, " Reg. context: %s\n", ast_strlen_zero(regcontext) ? "(not set)" : regcontext);
ast_cli(fd, " Caller ID: %s\n", default_callerid);
ast_cli(fd, " From: Domain: %s\n", default_fromdomain);
ast_cli(fd, " Record SIP history: %s\n", recordhistory ? "On" : "Off");
ast_cli(fd, " Call Events: %s\n", callevents ? "On" : "Off");
- ast_cli(fd, " IP ToS: 0x%x\n", tos);
+ ast_cli(fd, " IP ToS: 0x%x\n", global_tos);
#ifdef OSP_SUPPORT
ast_cli(fd, " OSP Support: Yes\n");
#else
@@ -8301,8 +8307,8 @@
ast_cli(fd, " Use ClientCode: %s\n", ast_test_flag(&global_flags, SIP_USECLIENTCODE) ? "Yes" : "No");
ast_cli(fd, " Progress inband: %s\n", (ast_test_flag(&global_flags, SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER) ? "Never" : (ast_test_flag(&global_flags, SIP_PROG_INBAND) == SIP_PROG_INBAND_NO) ? "No" : "Yes" );
ast_cli(fd, " Language: %s\n", ast_strlen_zero(default_language) ? "(Defaults to English)" : default_language);
- ast_cli(fd, " Musicclass: %s\n", global_musicclass);
- ast_cli(fd, " Voice Mail Extension: %s\n", global_vmexten);
+ ast_cli(fd, " Musicclass: %s\n", default_musicclass);
+ ast_cli(fd, " Voice Mail Extension: %s\n", default_vmexten);
if (realtimepeers || realtimeusers) {
@@ -11945,7 +11951,7 @@
/* set default context */
strcpy(user->context, default_context);
strcpy(user->language, default_language);
- strcpy(user->musicclass, global_musicclass);
+ strcpy(user->musicclass, default_musicclass);
for (; v; v = v->next) {
if (handle_common_options(&userflags, &mask, v))
continue;
@@ -12027,7 +12033,7 @@
strcpy(peer->context, default_context);
strcpy(peer->subscribecontext, default_subscribecontext);
strcpy(peer->language, default_language);
- strcpy(peer->musicclass, global_musicclass);
+ strcpy(peer->musicclass, default_musicclass);
peer->addr.sin_port = htons(DEFAULT_SIP_PORT);
peer->addr.sin_family = AF_INET;
peer->capability = global_capability;
@@ -12097,9 +12103,9 @@
}
strcpy(peer->context, default_context);
strcpy(peer->subscribecontext, default_subscribecontext);
- strcpy(peer->vmexten, global_vmexten);
+ strcpy(peer->vmexten, default_vmexten);
strcpy(peer->language, default_language);
- strcpy(peer->musicclass, global_musicclass);
+ strcpy(peer->musicclass, default_musicclass);
ast_copy_flags(peer, &global_flags, SIP_USEREQPHONE);
peer->secret[0] = '\0';
peer->md5secret[0] = '\0';
@@ -12331,58 +12337,66 @@
return -1;
}
+ /* Clear all flags before setting default values */
+ ast_clear_flag(&global_flags, AST_FLAGS_ALL);
+
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&prefs, 0 , sizeof(prefs));
- ast_clear_flag(&global_flags_page2, SIP_PAGE2_DEBUG_CONFIG);
-
- /* Initialize some reasonable defaults at SIP reload */
+ outboundproxyip.sin_port = htons(DEFAULT_SIP_PORT);
+ outboundproxyip.sin_family = AF_INET; /* Type of address: IPv4 */
+ ourport = DEFAULT_SIP_PORT;
+ srvlookup = DEFAULT_SRVLOOKUP;
+ global_tos = DEFAULT_TOS;
+ externhost[0] = '\0'; /* External host name (for behind NAT DynDNS support) */
+ externexpire = 0; /* Expiration for DNS re-issuing */
+ externrefresh = 10;
+ memset(&outboundproxyip, 0, sizeof(outboundproxyip));
+
+ /* Reset channel settings to default before re-configuring */
+ allow_external_domains = DEFAULT_ALLOW_EXT_DOM; /* Allow external invites */
+ regcontext[0] = '\0';
+ expiry = DEFAULT_EXPIRY;
+ global_notifyringing = DEFAULT_NOTIFYRINGING;
+ ast_copy_string(global_useragent, DEFAULT_USERAGENT, sizeof(global_useragent));
+ ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime));
+ ast_copy_string(global_realm, DEFAULT_REALM, sizeof(global_realm));
+ ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid));
+ global_videosupport = DEFAULT_VIDEOSUPPORT;
+ compactheaders = DEFAULT_COMPACTHEADERS;
+ global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
+ global_regattempts_max = 0;
+ pedanticsipchecking = DEFAULT_PEDANTIC;
+ global_mwitime = DEFAULT_MWITIME;
+ autocreatepeer = DEFAULT_AUTOCREATEPEER;
+ global_allowguest = DEFAULT_ALLOWGUEST;
+ global_rtptimeout = 0;
+ global_rtpholdtimeout = 0;
+ global_rtpkeepalive = 0;
+ ast_set_flag(&global_flags_page2, SIP_PAGE2_RTUPDATE);
+
+ /* Initialize some reasonable defaults at SIP reload (used both for channel and as default for peers and users */
ast_copy_string(default_context, DEFAULT_CONTEXT, sizeof(default_context));
default_subscribecontext[0] = '\0';
default_language[0] = '\0';
default_fromdomain[0] = '\0';
- default_qualify = 0;
- allow_external_domains = 1; /* Allow external invites */
- externhost[0] = '\0';
- externexpire = 0;
- externrefresh = 10;
- ast_copy_string(default_useragent, DEFAULT_USERAGENT, sizeof(default_useragent));
- ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime));
- global_notifyringing = 1;
- ast_copy_string(global_realm, DEFAULT_REALM, sizeof(global_realm));
- ast_copy_string(global_musicclass, "default", sizeof(global_musicclass));
- ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid));
- memset(&outboundproxyip, 0, sizeof(outboundproxyip));
- outboundproxyip.sin_port = htons(DEFAULT_SIP_PORT);
- outboundproxyip.sin_family = AF_INET; /* Type of address: IPv4 */
- videosupport = 0;
- compactheaders = 0;
- dumphistory = 0;
- recordhistory = 0;
+ default_qualify = DEFAULT_QUALIFY;
+ ast_copy_string(default_musicclass, DEFAULT_MUSICCLASS, sizeof(default_musicclass));
+ ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
+ ast_set_flag(&global_flags, SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
+ ast_set_flag(&global_flags, SIP_NAT_RFC3581); /*!< NAT support if requested by device with rport */
+ ast_set_flag(&global_flags, SIP_CAN_REINVITE); /*!< Allow re-invites */
+
+ /* Debugging settings, always default to off */
+ dumphistory = FALSE;
+ recordhistory = FALSE;
+ ast_clear_flag(&global_flags_page2, SIP_PAGE2_DEBUG_CONFIG);
+
+ /* Misc settings for the channel */
relaxdtmf = 0;
callevents = 0;
- ourport = DEFAULT_SIP_PORT;
- global_rtptimeout = 0;
- global_rtpholdtimeout = 0;
- global_rtpkeepalive = 0;
- pedanticsipchecking = 0;
- global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
- global_regattempts_max = 0;
- ast_clear_flag(&global_flags, AST_FLAGS_ALL);
- ast_set_flag(&global_flags, SIP_DTMF_RFC2833);
- ast_set_flag(&global_flags, SIP_NAT_RFC3581);
- ast_set_flag(&global_flags, SIP_CAN_REINVITE);
- ast_set_flag(&global_flags_page2, SIP_PAGE2_RTUPDATE);
- global_mwitime = DEFAULT_MWITIME;
- strcpy(global_vmexten, DEFAULT_VMEXTEN);
- srvlookup = 0;
- autocreatepeer = 0;
- regcontext[0] = '\0';
- tos = 0;
- expiry = DEFAULT_EXPIRY;
- global_allowguest = 1;
/* Read the [general] config section of sip.conf (or from realtime config) */
for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
@@ -12395,9 +12409,8 @@
} else if (!strcasecmp(v->name, "realm")) {
ast_copy_string(global_realm, v->value, sizeof(global_realm));
} else if (!strcasecmp(v->name, "useragent")) {
- ast_copy_string(default_useragent, v->value, sizeof(default_useragent));
- ast_log(LOG_DEBUG, "Setting User Agent Name to %s\n",
- default_useragent);
+ ast_copy_string(global_useragent, v->value, sizeof(global_useragent));
+ ast_log(LOG_DEBUG, "Setting SIP channel User-Agent Name to %s\n", global_useragent);
} else if (!strcasecmp(v->name, "rtcachefriends")) {
ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_RTCACHEFRIENDS);
} else if (!strcasecmp(v->name, "rtupdate")) {
@@ -12421,7 +12434,7 @@
global_mwitime = DEFAULT_MWITIME;
}
} else if (!strcasecmp(v->name, "vmexten")) {
- ast_copy_string(global_vmexten, v->value, sizeof(global_vmexten));
+ ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
} else if (!strcasecmp(v->name, "rtptimeout")) {
if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
@@ -12438,7 +12451,7 @@
global_rtpkeepalive = 0;
}
} else if (!strcasecmp(v->name, "videosupport")) {
- videosupport = ast_true(v->value);
+ global_videosupport = ast_true(v->value);
} else if (!strcasecmp(v->name, "compactheaders")) {
compactheaders = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifymimetype")) {
@@ -12446,7 +12459,7 @@
} else if (!strcasecmp(v->name, "notifyringing")) {
global_notifyringing = ast_true(v->value);
} else if (!strcasecmp(v->name, "musicclass") || !strcasecmp(v->name, "musiconhold")) {
- ast_copy_string(global_musicclass, v->value, sizeof(global_musicclass));
+ ast_copy_string(default_musicclass, v->value, sizeof(default_musicclass));
} else if (!strcasecmp(v->name, "language")) {
ast_copy_string(default_language, v->value, sizeof(default_language));
} else if (!strcasecmp(v->name, "regcontext")) {
@@ -12552,7 +12565,7 @@
} else if (!strcasecmp(v->name, "register")) {
sip_register(v->value, v->lineno);
} else if (!strcasecmp(v->name, "tos")) {
- if (ast_str2tos(v->value, &tos))
+ if (ast_str2tos(v->value, &global_tos))
ast_log(LOG_WARNING, "Invalid tos value at line %d, should be 'lowdelay', 'throughput', 'reliability', 'mincost', or 'none'\n", v->lineno);
} else if (!strcasecmp(v->name, "bindport")) {
if (sscanf(v->value, "%d", &ourport) == 1) {
@@ -12660,10 +12673,10 @@
if (option_verbose > 1) {
ast_verbose(VERBOSE_PREFIX_2 "SIP Listening on %s:%d\n",
ast_inet_ntoa(iabuf, sizeof(iabuf), bindaddr.sin_addr), ntohs(bindaddr.sin_port));
- ast_verbose(VERBOSE_PREFIX_2 "Using TOS bits %d\n", tos);
+ ast_verbose(VERBOSE_PREFIX_2 "Using TOS bits %d\n", global_tos);
}
- if (setsockopt(sipsock, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))
- ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
+ if (setsockopt(sipsock, IPPROTO_IP, IP_TOS, &global_tos, sizeof(global_tos)))
+ ast_log(LOG_WARNING, "Unable to set TOS to %d\n", global_tos);
}
}
}
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