- 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 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8514 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej
parent
4df94f3769
commit
6a63b0622a
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@ -92,8 +92,12 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/astosp.h"
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#endif
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#ifndef DEFAULT_USERAGENT
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#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
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#ifndef FALSE
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#define FALSE 0
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#endif
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#ifndef TRUE
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#define TRUE 1
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#endif
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#define VIDEO_CODEC_MASK 0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
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@ -111,19 +115,21 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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/* guard limit must be larger than guard secs */
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/* guard min must be < 1000, and should be >= 250 */
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#define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
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#define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of
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EXPIRY_GUARD_SECS */
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#define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
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GUARD_PCT turns out to be lower than this, it
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will use this time instead.
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This is in milliseconds. */
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#define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
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below EXPIRY_GUARD_LIMIT */
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#define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
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#define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of
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EXPIRY_GUARD_SECS */
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#define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
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GUARD_PCT turns out to be lower than this, it
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will use this time instead.
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This is in milliseconds. */
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#define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
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below EXPIRY_GUARD_LIMIT */
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#define DEFAULT_EXPIRY 900 /*!< Expire slowly */
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static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
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static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
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static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
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static int expiry = DEFAULT_EXPIRY;
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#ifndef MAX
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#define MAX(a,b) ((a) > (b) ? (a) : (b))
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@ -133,14 +139,17 @@ static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
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#define DEFAULT_MAXMS 2000 /*!< Must be faster than 2 seconds by default */
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#define DEFAULT_FREQ_OK 60 * 1000 /*!< How often to check for the host to be up */
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#define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< How often to check, if the host is down... */
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#define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
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#define DEFAULT_FREQ_OK 60 * 1000 /*!< Qualification: How often to check for the host to be up */
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#define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< Qualification: How often to check, if the host is down... */
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#define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
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#define MAX_RETRANS 6 /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
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#define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
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#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
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#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
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static const char desc[] = "Session Initiation Protocol (SIP)";
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static const char channeltype[] = "SIP";
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@ -326,76 +335,90 @@ static const struct cfsip_options {
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/*! \brief SIP Extensions we support */
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#define SUPPORTED_EXTENSIONS "replaces"
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#define DEFAULT_SIP_PORT 5060 /*!< From RFC 3261 (former 2543) */
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#define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */
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static char default_useragent[AST_MAX_EXTENSION] = DEFAULT_USERAGENT;
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/* Default values, set and reset in reload_config before reading configuration */
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/* These are default values in the source. There are other recommended values in the
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sip.conf.sample for new installations. These may differ to keep backwards compatibility,
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yet encouraging new behaviour on new installations
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*/
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#define DEFAULT_SIP_PORT 5060 /*!< From RFC 3261 (former 2543) */
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#define DEFAULT_CONTEXT "default"
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#define DEFAULT_MUSICCLASS "default"
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#define DEFAULT_VMEXTEN "asterisk"
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#define DEFAULT_CALLERID "asterisk"
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#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
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#define DEFAULT_MWITIME 10
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#define DEFAULT_ALLOWGUEST TRUE
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#define DEFAULT_VIDEOSUPPORT FALSE
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#define DEFAULT_SRVLOOKUP FALSE /*!< Recommended setting is ON */
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#define DEFAULT_COMPACTHEADERS FALSE
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#define DEFAULT_TOS FALSE
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#define DEFAULT_ALLOW_EXT_DOM TRUE
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#define DEFAULT_REALM "asterisk"
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#define DEFAULT_NOTIFYRINGING TRUE
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#define DEFAULT_PEDANTIC FALSE
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#define DEFAULT_AUTOCREATEPEER FALSE
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#define DEFAULT_QUALIFY FALSE
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#ifndef DEFAULT_USERAGENT
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#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
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#endif
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#define DEFAULT_CONTEXT "default"
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/* Default setttings are used as a channel setting and as a default when
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configuring devices */
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static char default_context[AST_MAX_CONTEXT];
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static char default_subscribecontext[AST_MAX_CONTEXT];
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#define DEFAULT_VMEXTEN "asterisk"
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static char global_vmexten[AST_MAX_EXTENSION];
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static char default_language[MAX_LANGUAGE];
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#define DEFAULT_CALLERID "asterisk"
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static char default_callerid[AST_MAX_EXTENSION];
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static char default_fromdomain[AST_MAX_EXTENSION];
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#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
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static char default_notifymime[AST_MAX_EXTENSION];
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static int global_notifyringing; /*!< Send notifications on ringing */
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static int default_qualify; /*!< Default Qualify= setting */
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static char default_vmexten[AST_MAX_EXTENSION];
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static char default_musicclass[MAX_MUSICCLASS]; /*!< Global music on hold class */
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/* Global settings only apply to the channel */
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static int global_notifyringing; /*!< Send notifications on ringing */
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static int srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
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static int pedanticsipchecking; /*!< Extra checking ? Default off */
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static int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
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static int relaxdtmf; /*!< Relax DTMF */
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static int global_rtptimeout; /*!< Time out call if no RTP */
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static int global_rtpholdtimeout;
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static int global_rtpkeepalive; /*!< Send RTP keepalives */
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static int global_reg_timeout;
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static int global_regattempts_max; /*!< Registration attempts before giving up */
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static int global_allowguest = 1; /*!< allow unauthenticated users/peers to connect? */
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#define DEFAULT_MWITIME 10
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static int global_allowguest; /*!< allow unauthenticated users/peers to connect? */
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static int global_mwitime; /*!< Time between MWI checks for peers */
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static int global_tos; /*!< IP Type of service */
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static int global_videosupport; /*!< Videosupport on or off */
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static int compactheaders; /*!< send compact sip headers */
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static int recordhistory; /*!< Record SIP history. Off by default */
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static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
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static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */
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static char regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
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static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
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static int allow_external_domains; /*!< Accept calls to external SIP domains? */
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static int tos = 0;
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static int videosupport = 0;
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static int compactheaders = 0; /*!< send compact sip headers */
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/*! \brief Codecs that we support by default: */
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static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
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static int noncodeccapability = AST_RTP_DTMF;
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/* Object counters */
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static int suserobjs = 0;
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static int ruserobjs = 0;
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static int speerobjs = 0;
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static int rpeerobjs = 0;
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static int apeerobjs = 0;
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static int regobjs = 0;
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static int suserobjs = 0; /*!< Static users */
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static int ruserobjs = 0; /*!< Realtime users */
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static int speerobjs = 0; /*!< Statis peers */
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static int rpeerobjs = 0; /*!< Realtime peers */
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static int apeerobjs = 0; /*!< Autocreated peer objects */
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static int regobjs = 0; /*!< Registry objects */
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static struct ast_flags global_flags = {0}; /*!< global SIP_ flags */
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static struct ast_flags global_flags_page2 = {0}; /*!< more global SIP_ flags */
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static int usecnt =0;
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AST_MUTEX_DEFINE_STATIC(usecnt_lock);
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AST_MUTEX_DEFINE_STATIC(rand_lock);
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AST_MUTEX_DEFINE_STATIC(rand_lock); /*!< Lock for thread-safe random generator */
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/*! \brief Protect the SIP dialog list (of sip_pvt's) */
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AST_MUTEX_DEFINE_STATIC(iflock);
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@ -412,32 +435,16 @@ static pthread_t monitor_thread = AST_PTHREADT_NULL;
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static int restart_monitor(void);
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/*! \brief Codecs that we support by default: */
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static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
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static int noncodeccapability = AST_RTP_DTMF;
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static struct in_addr __ourip;
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static struct sockaddr_in outboundproxyip;
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static int ourport;
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static struct sockaddr_in debugaddr;
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static int recordhistory; /*!< Record SIP history. Off by default */
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static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
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static char global_musicclass[MAX_MUSICCLASS]; /*!< Global music on hold class */
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#define DEFAULT_REALM "asterisk"
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static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */
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static char regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
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#define DEFAULT_EXPIRY 900 /*!< Expire slowly */
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static int expiry = DEFAULT_EXPIRY;
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static struct sched_context *sched;
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static struct io_context *io;
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#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
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#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
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#define DEC_CALL_LIMIT 0
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#define INC_CALL_LIMIT 1
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@ -493,7 +500,6 @@ struct domain {
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static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */
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int allow_external_domains; /*!< Accept calls to external SIP domains? */
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/*! \brief sip_history: Structure for saving transactions within a SIP dialog */
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struct sip_history {
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@ -3129,10 +3135,10 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
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if (sip_methods[intended_method].need_rtp) {
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p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
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if (videosupport)
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if (global_videosupport)
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p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
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if (!p->rtp || (videosupport && !p->vrtp)) {
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ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n", videosupport ? "and video" : "", strerror(errno));
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if (!p->rtp || (global_videosupport && !p->vrtp)) {
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ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n", global_videosupport ? "and video" : "", strerror(errno));
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ast_mutex_destroy(&p->lock);
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if (p->chanvars) {
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ast_variables_destroy(p->chanvars);
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@ -3141,9 +3147,9 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
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free(p);
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return NULL;
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}
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ast_rtp_settos(p->rtp, tos);
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ast_rtp_settos(p->rtp, global_tos);
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if (p->vrtp)
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ast_rtp_settos(p->vrtp, tos);
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ast_rtp_settos(p->vrtp, global_tos);
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p->rtptimeout = global_rtptimeout;
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p->rtpholdtimeout = global_rtpholdtimeout;
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p->rtpkeepalive = global_rtpkeepalive;
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@ -3168,7 +3174,7 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
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ast_string_field_set(p, callid, callid);
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ast_copy_flags(p, &global_flags, SIP_FLAGS_TO_COPY);
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/* Assign default music on hold class */
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ast_string_field_set(p, musicclass, global_musicclass);
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ast_string_field_set(p, musicclass, default_musicclass);
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p->capability = global_capability;
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if ((ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833) || (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_AUTO))
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p->noncodeccapability |= AST_RTP_DTMF;
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@ -4071,7 +4077,7 @@ static int respprep(struct sip_request *resp, struct sip_pvt *p, char *msg, stru
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add_header(resp, "To", ot);
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copy_header(resp, req, "Call-ID");
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copy_header(resp, req, "CSeq");
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add_header(resp, "User-Agent", default_useragent);
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add_header(resp, "User-Agent", global_useragent);
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add_header(resp, "Allow", ALLOWED_METHODS);
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if (msg[0] == '2' && (p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER)) {
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/* For registration responses, we also need expiry and
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@ -4187,7 +4193,7 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, in
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copy_header(req, orig, "Call-ID");
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add_header(req, "CSeq", tmp);
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add_header(req, "User-Agent", default_useragent);
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add_header(req, "User-Agent", global_useragent);
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add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
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if (!ast_strlen_zero(p->rpid))
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@ -4506,7 +4512,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
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}
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/* Now send any other common codecs, and non-codec formats: */
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for (x = 1; x <= ((videosupport && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) {
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for (x = 1; x <= ((global_videosupport && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) {
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if (!(capability & x))
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continue;
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@ -4916,7 +4922,7 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
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add_header(req, "Contact", p->our_contact);
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add_header(req, "Call-ID", p->callid);
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add_header(req, "CSeq", tmp);
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add_header(req, "User-Agent", default_useragent);
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add_header(req, "User-Agent", global_useragent);
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add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
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if (!ast_strlen_zero(p->rpid))
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add_header(req, "Remote-Party-ID", p->rpid);
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@ -5205,7 +5211,7 @@ static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs,
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add_header(&req, "Content-Type", default_notifymime);
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ast_build_string(&t, &maxbytes, "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no");
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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);
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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);
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ast_build_string(&t, &maxbytes, "Voice-Message: %d/%d (0/0)\r\n", newmsgs, oldmsgs);
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if (t > tmp + sizeof(tmp))
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@ -5511,7 +5517,7 @@ static int transmit_register(struct sip_registry *r, int sipmethod, char *auth,
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add_header(&req, "To", to);
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add_header(&req, "Call-ID", p->callid);
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add_header(&req, "CSeq", tmp);
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add_header(&req, "User-Agent", default_useragent);
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add_header(&req, "User-Agent", global_useragent);
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add_header(&req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
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@ -8247,7 +8253,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
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ast_cli(fd, "----------------\n");
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ast_cli(fd, " SIP Port: %d\n", ntohs(bindaddr.sin_port));
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ast_cli(fd, " Bindaddress: %s\n", ast_inet_ntoa(tmp, sizeof(tmp), bindaddr.sin_addr));
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ast_cli(fd, " Videosupport: %s\n", videosupport ? "Yes" : "No");
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ast_cli(fd, " Videosupport: %s\n", global_videosupport ? "Yes" : "No");
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ast_cli(fd, " AutoCreatePeer: %s\n", autocreatepeer ? "Yes" : "No");
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ast_cli(fd, " Allow unknown access: %s\n", global_allowguest ? "Yes" : "No");
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ast_cli(fd, " Promsic. redir: %s\n", ast_test_flag(&global_flags, SIP_PROMISCREDIR) ? "Yes" : "No");
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@ -8256,14 +8262,14 @@ static int sip_show_settings(int fd, int argc, char *argv[])
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ast_cli(fd, " URI user is phone no: %s\n", ast_test_flag(&global_flags, SIP_USEREQPHONE) ? "Yes" : "No");
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ast_cli(fd, " Our auth realm %s\n", global_realm);
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ast_cli(fd, " Realm. auth: %s\n", authl ? "Yes": "No");
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ast_cli(fd, " User Agent: %s\n", default_useragent);
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ast_cli(fd, " User Agent: %s\n", global_useragent);
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ast_cli(fd, " MWI checking interval: %d secs\n", global_mwitime);
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ast_cli(fd, " Reg. context: %s\n", ast_strlen_zero(regcontext) ? "(not set)" : regcontext);
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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);
|
||||
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));
|
||||
|
||||
/* Initialize some reasonable defaults at SIP reload */
|
||||
/* 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 @@ 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);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
|
|
|||
Reference in New Issue