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- Adding doxygen comments

Browse Source 
- 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



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Olle Johansson 20 years ago
parent 428efda5a9
commit b90a5b0ebd
1 changed files with 153 additions and 140 deletions
Show Stats Download Patch File Download Diff File

293
channels/chan_sip.c
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@ -92,8 +92,12 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#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 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
/* 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))
@ -133,14 +139,17 @@ static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
#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)";
static const char channeltype[] = "SIP";
@ -326,76 +335,90 @@ static const struct cfsip_options {
/*! \brief SIP Extensions we support */
#define SUPPORTED_EXTENSIONS "replaces"
#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;
/* 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 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
#define DEFAULT_CONTEXT "default"
/* 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 global_notifyringing; /*!< Send notifications on ringing */
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 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 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? */
static int tos = 0;
static int videosupport = 0;
static int compactheaders = 0; /*!< send compact sip headers */
/*! \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 pthread_t monitor_thread = AST_PTHREADT_NULL;
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 @@ struct domain {
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 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
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 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
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 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
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 @@ static int respprep(struct sip_request *resp, struct sip_pvt *p, char *msg, stru
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 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, in
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 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
}
/* 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 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
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 @@ static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs,
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 @@ static int transmit_register(struct sip_registry *r, int sipmethod, char *auth,
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 @@ static int sip_show_settings(int fd, int argc, char *argv[])
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 @@ static int sip_show_settings(int fd, int argc, char *argv[])
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 @@ static int sip_show_settings(int fd, int argc, char *argv[])
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 @@ static struct sip_user *build_user(const char *name, struct ast_variable *v, int
/* 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 @@ static struct sip_peer *temp_peer(const char *name)
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 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int
}
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 @@ static int reload_config(void)
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 */
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;
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;
ast_copy_string(default_useragent, DEFAULT_USERAGENT, sizeof(default_useragent));
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));
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;
relaxdtmf = 0;
callevents = 0;
ourport = DEFAULT_SIP_PORT;
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;
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;
/* 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 = 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;
/* 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 @@ static int reload_config(void)
} 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 @@ static int reload_config(void)
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 @@ static int reload_config(void)
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 @@ static int reload_config(void)
} 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 @@ static int reload_config(void)
} 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 @@ static int reload_config(void)
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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