chan_sip header file More...

#include "asterisk.h"
#include "asterisk/stringfields.h"
#include "asterisk/linkedlists.h"
#include "asterisk/strings.h"
#include "asterisk/tcptls.h"
#include "asterisk/test.h"
#include "asterisk/channel.h"
#include "asterisk/app.h"
#include "asterisk/indications.h"
#include "asterisk/security_events.h"
#include "asterisk/features.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/netsock2.h"
#include "asterisk/features_config.h"
#include "route.h"

Include dependency graph for sip.h:

This graph shows which files directly or indirectly include this file:

Go to the source code of this file.

Data Structures
struct	__show_chan_arg
	argument for the 'show channels\|subscriptions' callback. More...

struct	_map_x_s
	generic struct to map between strings and integers. Fill it with x-s pairs, terminate with an entry with s = NULL; Then you can call map_x_s(...) to map an integer to a string, and map_s_x() for the string -> integer mapping. More...

struct	cc_epa_entry
	Instance data for a Call completion EPA entry. More...

struct	cfsip_options
	List of well-known SIP options. If we get this in a require, we should check the list and answer accordingly. More...

struct	digestkeys

struct	domain
	Domain data structure. More...

struct	epa_backend
	backend for an event publication agent More...

struct	epa_static_data

struct	offered_media
	Structure for remembering offered media in an INVITE, to make sure we reply to all media streams. More...

struct	sip_pvt::request_queue

struct	sip_auth
	sip_auth: Credentials for authentication to other SIP services More...

struct	sip_auth_container
	Container of SIP authentication credentials. More...

struct	sip_cc_agent_pvt

struct	sip_epa_entry

struct	sip_esc_entry
	common ESC items for all event types More...

struct	sip_esc_publish_callbacks
	Callbacks for SIP ESCs. More...

struct	sip_history
	sip_history: Structure for saving transactions within a SIP dialog More...

struct	sip_invite_param
	Parameters to the transmit_invite function. More...

struct	sip_mailbox
	A peer's mailbox. More...

struct	sip_monitor_instance

struct	sip_msg_hdr

struct	sip_notify
	Struct to handle custom SIP notify requests. Dynamically allocated when needed. More...

struct	sip_peer
	Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host) More...

struct	sip_pkt
	sip packet - raw format for outbound packets that are sent or scheduled for transmission Packets are linked in a list, whose head is in the struct sip_pvt they belong to. Each packet holds a reference to the parent struct sip_pvt. This structure is allocated in __sip_reliable_xmit() and only for packets that require retransmissions. More...

struct	sip_proxy
	definition of a sip proxy server More...

struct	sip_pvt
	Structure used for each SIP dialog, ie. a call, a registration, a subscribe. Created and initialized by sip_alloc(), the descriptor goes into the list of descriptors (dialoglist). More...

struct	sip_refer
	Structure to handle SIP transfers. Dynamically allocated when needed. More...

struct	sip_registry
	Registrations with other SIP proxies. More...

struct	sip_request
	sip_request: The data grabbed from the UDP socket More...

struct	sip_settings
	a place to store all global settings for the sip channel driver More...

struct	sip_socket
	The SIP socket definition. More...

struct	sip_st_cfg
	Structure that encapsulates all attributes related to configuration of SIP Session-Timers feature on a per user/peer basis. More...

struct	sip_st_dlg
	Structure that encapsulates all attributes related to running SIP Session-Timers feature on a per dialog basis. More...

struct	sip_subscription_mwi
	Definition of an MWI subscription to another server. More...

struct	sip_threadinfo
	Definition of a thread that handles a socket. More...

struct	sip_via
	Structure to store Via information. More...

struct	t38properties
	T.38 channel settings (at some point we need to make this alloc'ed. More...

struct	tcptls_packet

Macros
#define	ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE"
	SIP Methods we support. More...

#define	CALLERID_UNKNOWN "Anonymous"

#define	DEC_CALL_LIMIT 0

#define	DEC_CALL_RINGING 2

#define	DEFAULT_AUTHLIMIT 100

#define	DEFAULT_AUTHTIMEOUT 30

#define	DEFAULT_DEFAULT_EXPIRY 120

#define	DEFAULT_EXPIRY 900

#define	DEFAULT_FREQ_NOTOK 10 * 1000

#define	DEFAULT_MAX_EXPIRY 3600

#define	DEFAULT_MAX_FORWARDS 70

#define	DEFAULT_MAX_SE 1800

#define	DEFAULT_MAXMS 2000

#define	DEFAULT_MIN_EXPIRY 60

#define	DEFAULT_MIN_SE 90

#define	DEFAULT_MWI_EXPIRY 3600

#define	DEFAULT_QUALIFY_GAP 100

#define	DEFAULT_QUALIFY_PEERS 1

#define	DEFAULT_QUALIFYFREQ 60 * 1000

#define	DEFAULT_REGISTRATION_TIMEOUT 20

#define	DEFAULT_RETRANS 1000

#define	DEFAULT_TIMER_T1 500

#define	DEFAULT_TRANS_TIMEOUT -1

#define	EXPIRY_GUARD_LIMIT 30

#define	EXPIRY_GUARD_MIN 500

#define	EXPIRY_GUARD_PCT 0.20

#define	EXPIRY_GUARD_SECS 15

#define	FALSE 0

#define	FINDALLDEVICES (FINDUSERS \| FINDPEERS)

#define	FINDPEERS (1 << 1)

#define	FINDUSERS (1 << 0)

#define	FROMDOMAIN_INVALID "anonymous.invalid"

#define	INC_CALL_LIMIT 1

#define	INC_CALL_RINGING 3

#define	INITIAL_CSEQ 101

#define	MAX_AUTHTRIES 3

#define	MAX_HISTORY_ENTRIES 50

#define	NO_RTP 0

#define	NOT_SUPPORTED 0

#define	OFFERED_MEDIA_COUNT 4
	The number of media types in enum media_type below. More...

#define	PROVIS_KEEPALIVE_TIMEOUT 60000

#define	RTP 1

#define	SDP_MAX_RTPMAP_CODECS 32

#define	SIP_MAX_HEADERS 64

#define	SIP_MAX_LINES 256

#define	SIP_MAX_PACKET_SIZE 20480

#define	SIP_MIN_PACKET 4096

#define	SIP_OPT_100REL (1 << 1)

#define	SIP_OPT_EARLY_SESSION (1 << 3)

#define	SIP_OPT_EVENTLIST (1 << 11)

#define	SIP_OPT_FROMCHANGE (1 << 17)

#define	SIP_OPT_GRUU (1 << 12)

#define	SIP_OPT_HISTINFO (1 << 15)

#define	SIP_OPT_JOIN (1 << 4)

#define	SIP_OPT_NOREFERSUB (1 << 14)

#define	SIP_OPT_OUTBOUND (1 << 20)

#define	SIP_OPT_PATH (1 << 5)

#define	SIP_OPT_PRECONDITION (1 << 7)

#define	SIP_OPT_PREF (1 << 6)

#define	SIP_OPT_PRIVACY (1 << 8)

#define	SIP_OPT_RECLISTINV (1 << 18)

#define	SIP_OPT_RECLISTSUB (1 << 19)

#define	SIP_OPT_REPLACES (1 << 0)

#define	SIP_OPT_RESPRIORITY (1 << 16)

#define	SIP_OPT_SDP_ANAT (1 << 9)

#define	SIP_OPT_SEC_AGREE (1 << 10)

#define	SIP_OPT_TARGET_DIALOG (1 << 13)

#define	SIP_OPT_TIMER (1 << 2)

#define	SIP_OPT_UNKNOWN (1 << 21)

#define	SIP_RESERVED ";/?:@&=+$,# "

#define	SIP_TRANS_TIMEOUT 64 * DEFAULT_TIMER_T1

#define	SIPBUFSIZE 512

#define	STANDARD_SIP_PORT 5060
	Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS. More...

#define	STANDARD_TLS_PORT 5061
	Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS. More...

#define	SUPPORTED 1

#define	TRUE 1

#define	XMIT_ERROR -2

DefaultValues Default values, set and reset in reload_config before reading configuration
Note in many SIP headers, absence of a port number implies port 5060, and this is why we cannot change the above constant. There is a limited number of places in asterisk where we could, in principle, use a different "default" port number, but we do not support this feature at the moment. You can run Asterisk with SIP on a different port with a configuration option. If you change this value in the source code, the signalling will be incorrect. 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_CONTEXT "default"

#define	DEFAULT_RECORD_FEATURE "automon"

#define	DEFAULT_MOHINTERPRET "default"

#define	DEFAULT_MOHSUGGEST ""

#define	DEFAULT_VMEXTEN "asterisk"

#define	DEFAULT_CALLERID "asterisk"

#define	DEFAULT_MWI_FROM ""

#define	DEFAULT_NOTIFYMIME "application/simple-message-summary"

#define	DEFAULT_ALLOWGUEST TRUE

#define	DEFAULT_RTPKEEPALIVE 0

#define	DEFAULT_CALLCOUNTER FALSE

#define	DEFAULT_SRVLOOKUP TRUE

#define	DEFAULT_COMPACTHEADERS FALSE

#define	DEFAULT_TOS_SIP 0

#define	DEFAULT_TOS_AUDIO 0

#define	DEFAULT_TOS_VIDEO 0

#define	DEFAULT_TOS_TEXT 0

#define	DEFAULT_COS_SIP 4

#define	DEFAULT_COS_AUDIO 5

#define	DEFAULT_COS_VIDEO 6

#define	DEFAULT_COS_TEXT 5

#define	DEFAULT_ALLOW_EXT_DOM TRUE

#define	DEFAULT_REALM "asterisk"

#define	DEFAULT_DOMAINSASREALM FALSE

#define	DEFAULT_NOTIFYRINGING NOTIFYRINGING_ENABLED

#define	DEFAULT_NOTIFYCID DISABLED

#define	DEFAULT_PEDANTIC TRUE

#define	DEFAULT_AUTOCREATEPEER AUTOPEERS_DISABLED

#define	DEFAULT_MATCHEXTERNADDRLOCALLY FALSE

#define	DEFAULT_QUALIFY FALSE

#define	DEFAULT_KEEPALIVE 0

#define	DEFAULT_KEEPALIVE_INTERVAL 60

#define	DEFAULT_ALWAYSAUTHREJECT TRUE

#define	DEFAULT_AUTH_OPTIONS FALSE

#define	DEFAULT_AUTH_MESSAGE TRUE

#define	DEFAULT_ACCEPT_OUTOFCALL_MESSAGE TRUE

#define	DEFAULT_REGEXTENONQUALIFY FALSE

#define	DEFAULT_LEGACY_USEROPTION_PARSING FALSE

#define	DEFAULT_SEND_DIVERSION TRUE

#define	DEFAULT_T1MIN 100

#define	DEFAULT_MAX_CALL_BITRATE (384)

#define	DEFAULT_USERAGENT "Asterisk PBX"

#define	DEFAULT_SDPSESSION "Asterisk PBX"

#define	DEFAULT_SDPOWNER "root"

#define	DEFAULT_ENGINE "asterisk"

#define	DEFAULT_STORE_SIP_CAUSE FALSE

SIPflags
Various flags for the flags field in the pvt structure Trying to sort these up (one or more of the following): D: Dialog P: Peer/user G: Global flag When flags are used by multiple structures, it is important that they have a common layout so it is easy to copy them.
#define	SIP_OUTGOING (1 << 0)

#define	SIP_OFFER_CC (1 << 1)

#define	SIP_RINGING (1 << 2)

#define	SIP_PROGRESS_SENT (1 << 3)

#define	SIP_NEEDREINVITE (1 << 4)

#define	SIP_PENDINGBYE (1 << 5)

#define	SIP_GOTREFER (1 << 6)

#define	SIP_CALL_LIMIT (1 << 7)

#define	SIP_INC_COUNT (1 << 8)

#define	SIP_INC_RINGING (1 << 9)

#define	SIP_DEFER_BYE_ON_TRANSFER (1 << 10)

#define	SIP_PROMISCREDIR (1 << 11)

#define	SIP_TRUSTRPID (1 << 12)

#define	SIP_USEREQPHONE (1 << 13)

#define	SIP_USECLIENTCODE (1 << 14)

#define	SIP_DTMF (7 << 15)

#define	SIP_DTMF_RFC2833 (0 << 15)

#define	SIP_DTMF_INBAND (1 << 15)

#define	SIP_DTMF_INFO (2 << 15)

#define	SIP_DTMF_AUTO (3 << 15)

#define	SIP_DTMF_SHORTINFO (4 << 15)

#define	SIP_NAT_FORCE_RPORT (1 << 18)

#define	SIP_NAT_RPORT_PRESENT (1 << 19)

#define	SIP_REINVITE (7 << 20)

#define	SIP_REINVITE_NONE (0 << 20)

#define	SIP_DIRECT_MEDIA (1 << 20)

#define	SIP_DIRECT_MEDIA_NAT (2 << 20)

#define	SIP_REINVITE_UPDATE (4 << 20)

#define	SIP_INSECURE (3 << 23)

#define	SIP_INSECURE_NONE (0 << 23)

#define	SIP_INSECURE_PORT (1 << 23)

#define	SIP_INSECURE_INVITE (1 << 24)

#define	SIP_PROG_INBAND (3 << 25)

#define	SIP_PROG_INBAND_NO (0 << 25)

#define	SIP_PROG_INBAND_NEVER (1 << 25)

#define	SIP_PROG_INBAND_YES (2 << 25)

#define	SIP_USEPATH (1 << 27)

#define	SIP_SENDRPID (3 << 29)

#define	SIP_SENDRPID_NO (0 << 29)

#define	SIP_SENDRPID_PAI (1 << 29)

#define	SIP_SENDRPID_RPID (2 << 29)

#define	SIP_G726_NONSTANDARD (1 << 31)

#define	SIP_FLAGS_TO_COPY
	Flags to copy from peer/user to dialog. More...

SIPflags2
a second page of flags (for flags[1]
#define	SIP_PAGE2_RTCACHEFRIENDS (1 << 0)

#define	SIP_PAGE2_RTAUTOCLEAR (1 << 1)

#define	SIP_PAGE2_RPID_UPDATE (1 << 2)

#define	SIP_PAGE2_Q850_REASON (1 << 3)

#define	SIP_PAGE2_SYMMETRICRTP (1 << 4)

#define	SIP_PAGE2_STATECHANGEQUEUE (1 << 5)

#define	SIP_PAGE2_CONNECTLINEUPDATE_PEND (1 << 6)

#define	SIP_PAGE2_RPID_IMMEDIATE (1 << 7)

#define	SIP_PAGE2_RPORT_PRESENT (1 << 8)

#define	SIP_PAGE2_PREFERRED_CODEC (1 << 9)

#define	SIP_PAGE2_VIDEOSUPPORT (1 << 10)

#define	SIP_PAGE2_TEXTSUPPORT (1 << 11)

#define	SIP_PAGE2_ALLOWSUBSCRIBE (1 << 12)

#define	SIP_PAGE2_ALLOWOVERLAP (3 << 13)

#define	SIP_PAGE2_ALLOWOVERLAP_NO (0 << 13)

#define	SIP_PAGE2_ALLOWOVERLAP_YES (1 << 13)

#define	SIP_PAGE2_ALLOWOVERLAP_DTMF (2 << 13)

#define	SIP_PAGE2_ALLOWOVERLAP_SPARE (3 << 13)

#define	SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 15)

#define	SIP_PAGE2_IGNORESDPVERSION (1 << 16)

#define	SIP_PAGE2_T38SUPPORT (3 << 17)

#define	SIP_PAGE2_T38SUPPORT_UDPTL (1 << 17)

#define	SIP_PAGE2_T38SUPPORT_UDPTL_FEC (2 << 17)

#define	SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY (3 << 17)

#define	SIP_PAGE2_CALL_ONHOLD (3 << 19)

#define	SIP_PAGE2_CALL_ONHOLD_ACTIVE (1 << 19)

#define	SIP_PAGE2_CALL_ONHOLD_ONEDIR (2 << 19)

#define	SIP_PAGE2_CALL_ONHOLD_INACTIVE (3 << 19)

#define	SIP_PAGE2_RFC2833_COMPENSATE (1 << 21)

#define	SIP_PAGE2_BUGGY_MWI (1 << 22)

#define	SIP_PAGE2_DIALOG_ESTABLISHED (1 << 23)

#define	SIP_PAGE2_FAX_DETECT (3 << 24)

#define	SIP_PAGE2_FAX_DETECT_CNG (1 << 24)

#define	SIP_PAGE2_FAX_DETECT_T38 (2 << 24)

#define	SIP_PAGE2_FAX_DETECT_BOTH (3 << 24)

#define	SIP_PAGE2_UDPTL_DESTINATION (1 << 26)

#define	SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 27)

#define	SIP_PAGE2_HAVEPEERCONTEXT (1 << 28) /< Are we associated with a configured peer context? /

#define	SIP_PAGE2_USE_SRTP (1 << 29)

#define	SIP_PAGE2_TRUST_ID_OUTBOUND (3 << 30)

#define	SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY (0 << 30)

#define	SIP_PAGE2_TRUST_ID_OUTBOUND_NO (1 << 30)

#define	SIP_PAGE2_TRUST_ID_OUTBOUND_YES (2 << 30)

#define	SIP_PAGE2_FLAGS_TO_COPY

#define	SIP_PAGE3_SNOM_AOC (1 << 0)

#define	SIP_PAGE3_SRTP_TAG_32 (1 << 1)

#define	SIP_PAGE3_NAT_AUTO_RPORT (1 << 2)

#define	SIP_PAGE3_NAT_AUTO_COMEDIA (1 << 3)

#define	SIP_PAGE3_DIRECT_MEDIA_OUTGOING (1 << 4)

#define	SIP_PAGE3_USE_AVPF (1 << 5)

#define	SIP_PAGE3_ICE_SUPPORT (1 << 6)

#define	SIP_PAGE3_IGNORE_PREFCAPS (1 << 7)

#define	SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL (1 << 8)

#define	SIP_PAGE3_FORCE_AVP (1 << 9)

#define	SIP_PAGE3_RTCP_MUX (1 << 10)

#define	SIP_PAGE3_FLAGS_TO_COPY

#define	CHECK_AUTH_BUF_INITLEN 256

Enumerations
enum	autocreatepeer_mode { AUTOPEERS_DISABLED = 0, AUTOPEERS_VOLATILE, AUTOPEERS_PERSIST }
	Automatic peer registration behavior. More...

enum	can_create_dialog { CAN_NOT_CREATE_DIALOG, CAN_CREATE_DIALOG, CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
	States whether a SIP message can create a dialog in Asterisk. More...

enum	check_auth_result { AUTH_DONT_KNOW = -100, AUTH_SUCCESSFUL = 0, AUTH_CHALLENGE_SENT = 1, AUTH_SECRET_FAILED = -1, AUTH_USERNAME_MISMATCH = -2, AUTH_NOT_FOUND = -3, AUTH_UNKNOWN_DOMAIN = -5, AUTH_PEER_NOT_DYNAMIC = -6, AUTH_ACL_FAILED = -7, AUTH_BAD_TRANSPORT = -8, AUTH_RTP_FAILED = -9, AUTH_SESSION_LIMIT = -10 }
	Authentication result from check_auth* functions. More...

enum	digest_keys { K_RESP, K_URI, K_USER, K_NONCE, K_LAST }

enum	domain_mode { SIP_DOMAIN_AUTO, SIP_DOMAIN_CONFIG }
	Modes for SIP domain handling in the PBX. More...

enum	inv_req_result { INV_REQ_SUCCESS = 11, INV_REQ_FAILED = 10, INV_REQ_ERROR = 9 }
	The results from handling an invite request. More...

enum	invitestates { INV_NONE = 0, INV_CALLING = 1, INV_PROCEEDING = 2, INV_EARLY_MEDIA = 3, INV_COMPLETED = 4, INV_CONFIRMED = 5, INV_TERMINATED = 6, INV_CANCELLED = 7 }
	States for the INVITE transaction, not the dialog. More...

enum	media_type { SDP_AUDIO, SDP_VIDEO, SDP_IMAGE, SDP_TEXT, SDP_UNKNOWN }
	Media types generate different "dummy answers" for not accepting the offer of a media stream. We need to add definitions for each RTP profile. Secure RTP is not the same as normal RTP and will require a new definition. More...

enum	notifycid_setting { DISABLED = 0, ENABLED = 1, IGNORE_CONTEXT = 2 }
	Settings for the 'notifycid' option, see sip.conf.sample for details. More...

enum	notifyringing_setting { NOTIFYRINGING_DISABLED = 0, NOTIFYRINGING_ENABLED = 1, NOTIFYRINGING_NOTINUSE = 2 }
	Setting for the 'notifyringing' option, see sip.conf.sample for details. More...

enum	parse_register_result { PARSE_REGISTER_DENIED, PARSE_REGISTER_FAILED, PARSE_REGISTER_UPDATE, PARSE_REGISTER_QUERY }
	Results from the parse_register() function. More...

enum	referstatus { REFER_IDLE, REFER_SENT, REFER_RECEIVED, REFER_CONFIRMED, REFER_ACCEPTED, REFER_RINGING, REFER_200OK, REFER_FAILED, REFER_NOAUTH }
	Parameters to know status of transfer. More...

enum	sip_auth_type { PROXY_AUTH = 407, WWW_AUTH = 401 }
	Authentication types - proxy or www authentication. More...

enum	sip_debug_e { sip_debug_none = 0, sip_debug_config = 1, sip_debug_console = 2 }
	debugging state We store separately the debugging requests from the config file and requests from the CLI. Debugging is enabled if either is set (which means that if sipdebug is set in the config file, we can only turn it off by reloading the config). More...

enum	sip_get_dest_result { SIP_GET_DEST_EXTEN_MATCHMORE = 1, SIP_GET_DEST_EXTEN_FOUND = 0, SIP_GET_DEST_EXTEN_NOT_FOUND = -1, SIP_GET_DEST_REFUSED = -2, SIP_GET_DEST_INVALID_URI = -3 }
	Result from get_destination function. More...

enum	sip_peer_type { SIP_TYPE_PEER = (1 << 0), SIP_TYPE_USER = (1 << 1) }

enum	sip_result { AST_SUCCESS = 0, AST_FAILURE = -1 }
	The result of a lot of functions. More...

enum	sip_tcptls_alert { TCPTLS_ALERT_DATA, TCPTLS_ALERT_STOP }

enum	sipmethod { SIP_UNKNOWN, SIP_RESPONSE, SIP_REGISTER, SIP_OPTIONS, SIP_NOTIFY, SIP_INVITE, SIP_ACK, SIP_PRACK, SIP_BYE, SIP_REFER, SIP_SUBSCRIBE, SIP_MESSAGE, SIP_UPDATE, SIP_INFO, SIP_CANCEL, SIP_PUBLISH, SIP_PING }
	SIP Request methods known by Asterisk. More...

enum	sipregistrystate { REG_STATE_UNREGISTERED = 0, REG_STATE_REGSENT, REG_STATE_AUTHSENT, REG_STATE_REGISTERED, REG_STATE_REJECTED, REG_STATE_TIMEOUT, REG_STATE_NOAUTH, REG_STATE_FAILED }
	States for outbound registrations (with register= lines in sip.conf. More...

enum	st_mode { SESSION_TIMER_MODE_INVALID = 0, SESSION_TIMER_MODE_ACCEPT, SESSION_TIMER_MODE_ORIGINATE, SESSION_TIMER_MODE_REFUSE }
	Modes in which Asterisk can be configured to run SIP Session-Timers. More...

enum	st_refresher { SESSION_TIMER_REFRESHER_AUTO, SESSION_TIMER_REFRESHER_US, SESSION_TIMER_REFRESHER_THEM }
	The entity playing the refresher role for Session-Timers. More...

enum	st_refresher_param { SESSION_TIMER_REFRESHER_PARAM_UNKNOWN, SESSION_TIMER_REFRESHER_PARAM_UAC, SESSION_TIMER_REFRESHER_PARAM_UAS }

enum	subscriptiontype { NONE = 0, XPIDF_XML, DIALOG_INFO_XML, CPIM_PIDF_XML, PIDF_XML, MWI_NOTIFICATION, CALL_COMPLETION }
	Type of subscription, based on the packages we do support, see subscription_types. More...

enum	t38_action_flag { SDP_T38_NONE = 0, SDP_T38_INITIATE, SDP_T38_ACCEPT }

enum	t38state { T38_DISABLED = 0, T38_LOCAL_REINVITE, T38_PEER_REINVITE, T38_ENABLED, T38_REJECTED }
	T38 States for a call. More...

enum	transfermodes { TRANSFER_OPENFORALL, TRANSFER_CLOSED }
	Authorization scheme for call transfers. More...

enum	xmittype { XMIT_CRITICAL = 2, XMIT_RELIABLE = 1, XMIT_UNRELIABLE = 0 }
	When sending a SIP message, we can send with a few options, depending on type of SIP request. UNRELIABLE is moslty used for responses to repeated requests, where the original response would be sent RELIABLE in an INVITE transaction. More...

GlobalSettings
Global settings apply to the channel (often settings you can change in the general section of sip.conf
#define	REQ_OFFSET_TO_STR(req, offset) (ast_str_buffer((req)->data) + ((req)->offset))

#define	sip_ref_peer(peer, tag) ao2_t_bump(peer, tag)

#define	sip_unref_peer(peer, tag) ({ ao2_t_cleanup(peer, tag); (NULL); })

enum	sip_mailbox_status { SIP_MAILBOX_STATUS_UNKNOWN = 0, SIP_MAILBOX_STATUS_EXISTING, SIP_MAILBOX_STATUS_NEW }

enum	sip_cc_publish_state { CC_CLOSED, CC_OPEN }
	The states that can be represented in a SIP call-completion PUBLISH. More...

enum	sip_cc_notify_state { CC_QUEUED, CC_READY }
	The states that can be represented in a SIP call-completion NOTIFY. More...

enum	sip_publish_type { SIP_PUBLISH_UNKNOWN, SIP_PUBLISH_INITIAL, SIP_PUBLISH_REFRESH, SIP_PUBLISH_MODIFY, SIP_PUBLISH_REMOVE, SIP_PUBLISH_UNKNOWN, SIP_PUBLISH_INITIAL, SIP_PUBLISH_REFRESH, SIP_PUBLISH_MODIFY, SIP_PUBLISH_REMOVE }
	The types of PUBLISH messages defined in RFC 3903. More...

typedef int(*const	esc_publish_callback) (struct sip_pvt , struct sip_request , struct event_state_compositor , struct sip_esc_entry )

static const struct cfsip_options	sip_options []

static struct ast_threadstorage	check_auth_buf = { .once = PTHREAD_ONCE_INIT , .key_init = __init_check_auth_buf , .custom_init = NULL , }

static void	__init_check_auth_buf (void)

struct sip_peer *	sip_find_peer (const char peer, struct ast_sockaddr addr, int realtime, int which_objects, int devstate_only, int transport)
	Locate device by name or ip address. More...

void	sip_auth_headers (enum sip_auth_type code, char header, char respheader)
	return the request and response header for a 401 or 407 code More...

const char *	sip_get_header (const struct sip_request req, const char name)
	Get header from SIP request. More...

const char *	sip_get_transport (enum ast_transport t)
	Return transport as string. More...

Detailed Description

chan_sip header file

Definition in file sip.h.

Macro Definition Documentation

◆ ALLOWED_METHODS

#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE"

SIP Methods we support.

Todo:: This string should be set dynamically. We only support REFER and SUBSCRIBE if we have allowsubscribe and allowrefer on in sip.conf.

Definition at line 173 of file sip.h.

Referenced by respprep(), transmit_invite(), transmit_notify_with_sipfrag(), transmit_reinvite_with_sdp(), and update_connectedline().

◆ CALLERID_UNKNOWN

#define CALLERID_UNKNOWN "Anonymous"

Definition at line 93 of file sip.h.

Referenced by initreqprep().

◆ CHECK_AUTH_BUF_INITLEN

#define CHECK_AUTH_BUF_INITLEN 256

Definition at line 401 of file sip.h.

Referenced by check_auth(), sip_report_security_event(), and transmit_fake_auth_response().

◆ DEC_CALL_LIMIT

#define DEC_CALL_LIMIT 0

Definition at line 127 of file sip.h.

Referenced by handle_request_cancel(), handle_request_invite(), handle_response_invite(), sip_hangup(), sip_pvt_dtor(), and update_call_counter().

◆ DEC_CALL_RINGING

#define DEC_CALL_RINGING 2

Definition at line 129 of file sip.h.

Referenced by handle_response_invite(), and update_call_counter().

◆ DEFAULT_ACCEPT_OUTOFCALL_MESSAGE

#define DEFAULT_ACCEPT_OUTOFCALL_MESSAGE TRUE

Definition at line 232 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_ALLOW_EXT_DOM

#define DEFAULT_ALLOW_EXT_DOM TRUE

Allow external domains

Definition at line 218 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_ALLOWGUEST

#define DEFAULT_ALLOWGUEST TRUE

Definition at line 205 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_ALWAYSAUTHREJECT

#define DEFAULT_ALWAYSAUTHREJECT TRUE

Don't reject authentication requests always

Definition at line 229 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_AUTH_MESSAGE

#define DEFAULT_AUTH_MESSAGE TRUE

Definition at line 231 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_AUTH_OPTIONS

#define DEFAULT_AUTH_OPTIONS FALSE

Definition at line 230 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_AUTHLIMIT

#define DEFAULT_AUTHLIMIT 100

Definition at line 69 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_AUTHTIMEOUT

#define DEFAULT_AUTHTIMEOUT 30

Definition at line 70 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_AUTOCREATEPEER

#define DEFAULT_AUTOCREATEPEER AUTOPEERS_DISABLED

Don't create peers automagically

Definition at line 224 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_CALLCOUNTER

#define DEFAULT_CALLCOUNTER FALSE

Do not enable call counters by default

Definition at line 207 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_CALLERID

#define DEFAULT_CALLERID "asterisk"

Default caller ID

Definition at line 202 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_COMPACTHEADERS

#define DEFAULT_COMPACTHEADERS FALSE

Send compact (one-character) SIP headers. Default off

Definition at line 209 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_CONTEXT

#define DEFAULT_CONTEXT "default"

The default context for [general] section as well as devices

Definition at line 197 of file sip.h.

◆ DEFAULT_COS_AUDIO

#define DEFAULT_COS_AUDIO 5

Level 2 class of service for audio media

Definition at line 215 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_COS_SIP

#define DEFAULT_COS_SIP 4

Level 2 class of service for SIP signalling

Definition at line 214 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_COS_TEXT

#define DEFAULT_COS_TEXT 5

Level 2 class of service for text media (T.140)

Definition at line 217 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_COS_VIDEO

#define DEFAULT_COS_VIDEO 6

Level 2 class of service for video media

Definition at line 216 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_DEFAULT_EXPIRY

#define DEFAULT_DEFAULT_EXPIRY 120

Definition at line 62 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_DOMAINSASREALM

#define DEFAULT_DOMAINSASREALM FALSE

Use the domain option to guess the realm for registration and invite requests

Definition at line 220 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_ENGINE

#define DEFAULT_ENGINE "asterisk"

Default RTP engine to use for sessions

Definition at line 242 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_EXPIRY

#define DEFAULT_EXPIRY 900

Expire slowly

Definition at line 88 of file sip.h.

◆ DEFAULT_FREQ_NOTOK

#define DEFAULT_FREQ_NOTOK 10 * 1000

Qualification: How often to check, if the host is down...

Definition at line 98 of file sip.h.

◆ DEFAULT_KEEPALIVE

#define DEFAULT_KEEPALIVE 0

Don't send keep alive packets

Definition at line 227 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_KEEPALIVE_INTERVAL

#define DEFAULT_KEEPALIVE_INTERVAL 60

Send keep alive packets at 60 second intervals

Definition at line 228 of file sip.h.

Referenced by build_peer(), and reload_config().

◆ DEFAULT_LEGACY_USEROPTION_PARSING

#define DEFAULT_LEGACY_USEROPTION_PARSING FALSE

Definition at line 234 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_MATCHEXTERNADDRLOCALLY

#define DEFAULT_MATCHEXTERNADDRLOCALLY FALSE

Match extern IP locally default setting

Definition at line 225 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_MAX_CALL_BITRATE

#define DEFAULT_MAX_CALL_BITRATE (384)

Max bitrate for video

Definition at line 237 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_MAX_EXPIRY

#define DEFAULT_MAX_EXPIRY 3600

Definition at line 64 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_MAX_FORWARDS

#define DEFAULT_MAX_FORWARDS 70

Definition at line 67 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_MAX_SE

#define DEFAULT_MAX_SE 1800

Session-Timer Default Session-Expires period (RFC 4028)

Definition at line 119 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_MAXMS

#define DEFAULT_MAXMS 2000

Qualification: Must be faster than 2 seconds by default

Definition at line 96 of file sip.h.

◆ DEFAULT_MIN_EXPIRY

#define DEFAULT_MIN_EXPIRY 60

Definition at line 63 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_MIN_SE

#define DEFAULT_MIN_SE 90

Session-Timer Default Min-SE period (RFC 4028)

Definition at line 120 of file sip.h.

Referenced by build_peer(), reload_config(), and transmit_invite().

◆ DEFAULT_MOHINTERPRET

#define DEFAULT_MOHINTERPRET "default"

The default music class

Definition at line 199 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_MOHSUGGEST

#define DEFAULT_MOHSUGGEST ""

Definition at line 200 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_MWI_EXPIRY

#define DEFAULT_MWI_EXPIRY 3600

Definition at line 65 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_MWI_FROM

#define DEFAULT_MWI_FROM ""

Definition at line 203 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_NOTIFYCID

#define DEFAULT_NOTIFYCID DISABLED

Include CID with ringing notifications

Definition at line 222 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_NOTIFYMIME

#define DEFAULT_NOTIFYMIME "application/simple-message-summary"

Definition at line 204 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_NOTIFYRINGING

#define DEFAULT_NOTIFYRINGING NOTIFYRINGING_ENABLED

Notify devicestate system on ringing state

Definition at line 221 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_PEDANTIC

#define DEFAULT_PEDANTIC TRUE

Follow SIP standards for dialog matching

Definition at line 223 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_QUALIFY

#define DEFAULT_QUALIFY FALSE

Don't monitor devices

Definition at line 226 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_QUALIFY_GAP

#define DEFAULT_QUALIFY_GAP 100

Definition at line 90 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_QUALIFY_PEERS

#define DEFAULT_QUALIFY_PEERS 1

Definition at line 91 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_QUALIFYFREQ

#define DEFAULT_QUALIFYFREQ 60 * 1000

Qualification: How often to check for the host to be up

Definition at line 97 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_REALM

#define DEFAULT_REALM "asterisk"

Realm for HTTP digest authentication

Definition at line 219 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_RECORD_FEATURE

#define DEFAULT_RECORD_FEATURE "automon"

The default feature specified for use with INFO

Definition at line 198 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_REGEXTENONQUALIFY

#define DEFAULT_REGEXTENONQUALIFY FALSE

Definition at line 233 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_REGISTRATION_TIMEOUT

#define DEFAULT_REGISTRATION_TIMEOUT 20

Definition at line 66 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_RETRANS

#define DEFAULT_RETRANS 1000

How frequently to retransmit Default: 2 * 500 ms in RFC 3261

Definition at line 100 of file sip.h.

◆ DEFAULT_RTPKEEPALIVE

#define DEFAULT_RTPKEEPALIVE 0

Default RTPkeepalive setting

Definition at line 206 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_SDPOWNER

#define DEFAULT_SDPOWNER "root"

Default SDP username field in (o=) header unless re-defined in sip.conf

Definition at line 241 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_SDPSESSION

#define DEFAULT_SDPSESSION "Asterisk PBX"

Default SDP session name, (s=) header unless re-defined in sip.conf

Definition at line 240 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_SEND_DIVERSION

#define DEFAULT_SEND_DIVERSION TRUE

Definition at line 235 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_SRVLOOKUP

#define DEFAULT_SRVLOOKUP TRUE

Recommended setting is ON

Definition at line 208 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_STORE_SIP_CAUSE

#define DEFAULT_STORE_SIP_CAUSE FALSE

Don't store HASH(SIP_CAUSE,<channel name>) for channels by default

Definition at line 243 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_T1MIN

#define DEFAULT_T1MIN 100

100 MS for minimal roundtrip time

Definition at line 236 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_TIMER_T1

#define DEFAULT_TIMER_T1 500

SIP timer T1 (according to RFC 3261)

Definition at line 101 of file sip.h.

Referenced by __sip_reliable_xmit(), and reload_config().

◆ DEFAULT_TOS_AUDIO

#define DEFAULT_TOS_AUDIO 0

Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions.

Definition at line 211 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_TOS_SIP

#define DEFAULT_TOS_SIP 0

Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions.

Definition at line 210 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_TOS_TEXT

#define DEFAULT_TOS_TEXT 0

Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions.

Definition at line 213 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_TOS_VIDEO

#define DEFAULT_TOS_VIDEO 0

Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions.

Definition at line 212 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_TRANS_TIMEOUT

#define DEFAULT_TRANS_TIMEOUT -1

Use default SIP transaction timeout

Definition at line 107 of file sip.h.

Referenced by __sip_autodestruct(), auto_congest(), check_auth(), check_pendings(), extensionstate_update(), handle_incoming(), handle_invite_replaces(), handle_request_bye(), handle_request_cancel(), handle_request_info(), handle_request_invite(), handle_request_notify(), handle_request_options(), handle_request_publish(), handle_request_register(), handle_response_invite(), receive_message(), sip_hangup(), sip_msg_send(), sip_send_mwi_to_peer(), transmit_fake_auth_response(), and transmit_publish().

◆ DEFAULT_USERAGENT

#define DEFAULT_USERAGENT "Asterisk PBX"

Default Useragent: header unless re-defined in sip.conf

Definition at line 239 of file sip.h.

Referenced by reload_config().

◆ DEFAULT_VMEXTEN

#define DEFAULT_VMEXTEN "asterisk"

Default voicemail extension

Definition at line 201 of file sip.h.

Referenced by reload_config().

◆ EXPIRY_GUARD_LIMIT

#define EXPIRY_GUARD_LIMIT 30

Below here, we use EXPIRY_GUARD_PCT instead of EXPIRY_GUARD_SECS

Definition at line 75 of file sip.h.

Referenced by handle_response_register().

◆ EXPIRY_GUARD_MIN

#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.

Definition at line 76 of file sip.h.

Referenced by handle_response_register().

◆ EXPIRY_GUARD_PCT

#define EXPIRY_GUARD_PCT 0.20

Percentage of expires timeout to use when below EXPIRY_GUARD_LIMIT

Definition at line 85 of file sip.h.

Referenced by handle_response_register().

◆ EXPIRY_GUARD_SECS

#define EXPIRY_GUARD_SECS 15

How long before expiry do we reregister

Definition at line 74 of file sip.h.

Referenced by handle_response_register().

◆ FALSE

#define FALSE 0

Definition at line 44 of file sip.h.

◆ FINDALLDEVICES

#define FINDALLDEVICES (FINDUSERS | FINDPEERS)

Definition at line 54 of file sip.h.

Referenced by check_peer_ok(), find_by_name(), sip_devicestate(), and sip_find_peer_full().

◆ FINDPEERS

#define FINDPEERS (1 << 1)

Definition at line 53 of file sip.h.

Referenced by _sip_qualify_peer(), _sip_show_peer(), check_peer_ok(), create_addr(), find_by_name(), function_sippeer(), handle_request_notify(), manager_sip_peer_status(), realtime_peer(), receive_message(), register_verify(), sip_do_debug_peer(), sip_find_peer_by_ip_and_exten(), sip_find_peer_full(), sip_report_security_event(), sip_unregister(), and transmit_register().

◆ FINDUSERS

#define FINDUSERS (1 << 0)

Definition at line 52 of file sip.h.

Referenced by check_peer_ok(), find_by_name(), realtime_peer(), sip_find_peer_full(), and sip_show_user().

◆ FROMDOMAIN_INVALID

#define FROMDOMAIN_INVALID "anonymous.invalid"

Definition at line 94 of file sip.h.

Referenced by initreqprep().

◆ INC_CALL_LIMIT

#define INC_CALL_LIMIT 1

Definition at line 128 of file sip.h.

Referenced by handle_request_invite(), and update_call_counter().

◆ INC_CALL_RINGING

#define INC_CALL_RINGING 3

Definition at line 130 of file sip.h.

Referenced by sip_call(), and update_call_counter().

◆ INITIAL_CSEQ

#define INITIAL_CSEQ 101

Our initial sip sequence number

Definition at line 117 of file sip.h.

Referenced by __sip_alloc(), AST_TEST_DEFINE(), sip_parse_register_line(), and transmit_response_using_temp().

◆ MAX_AUTHTRIES

#define MAX_AUTHTRIES 3

Try authentication three times, then fail

Definition at line 109 of file sip.h.

Referenced by do_message_auth(), handle_response(), handle_response_invite(), handle_response_publish(), handle_response_register(), and handle_response_update().

◆ MAX_HISTORY_ENTRIES

#define MAX_HISTORY_ENTRIES 50

Max entires in the history list for a sip_pvt

Definition at line 115 of file sip.h.

Referenced by append_history_va().

◆ NO_RTP

#define NO_RTP 0

Definition at line 125 of file sip.h.

◆ NOT_SUPPORTED

#define NOT_SUPPORTED 0

Definition at line 143 of file sip.h.

◆ OFFERED_MEDIA_COUNT

#define OFFERED_MEDIA_COUNT 4

The number of media types in enum media_type below.

Definition at line 484 of file sip.h.

◆ PROVIS_KEEPALIVE_TIMEOUT

#define PROVIS_KEEPALIVE_TIMEOUT 60000

How long to wait before retransmitting a provisional response (rfc 3261 13.3.1.1)

Definition at line 108 of file sip.h.

Referenced by __update_provisional_keepalive_full(), and send_provisional_keepalive_full().

◆ REQ_OFFSET_TO_STR

#define REQ_OFFSET_TO_STR	(	req,
		offset
	)	(ast_str_buffer((req)->data) + ((req)->offset))

◆ RTP

#define RTP 1

Definition at line 124 of file sip.h.

◆ SDP_MAX_RTPMAP_CODECS

#define SDP_MAX_RTPMAP_CODECS 32

Maximum number of codecs allowed in received SDP

Definition at line 122 of file sip.h.

Referenced by process_sdp_a_audio(), process_sdp_a_text(), and process_sdp_a_video().

◆ SIP_CALL_LIMIT

#define SIP_CALL_LIMIT (1 << 7)

D: Call limit enforced for this call

Definition at line 264 of file sip.h.

Referenced by check_peer_ok(), create_addr_from_peer(), and update_call_counter().

◆ SIP_DEFER_BYE_ON_TRANSFER

#define SIP_DEFER_BYE_ON_TRANSFER (1 << 10)

D: Do not hangup at first ast_hangup

Definition at line 267 of file sip.h.

Referenced by handle_request_refer(), local_attended_transfer(), sip_hangup(), and sip_set_rtp_peer().

◆ SIP_DIRECT_MEDIA

#define SIP_DIRECT_MEDIA (1 << 20)

DP: allow peers to be reinvited to send media directly p2p

Definition at line 289 of file sip.h.

Referenced by _sip_show_peer(), handle_common_options(), handle_incoming(), reload_config(), sip_allow_anyrtp_remote(), sip_get_rtp_peer(), sip_get_trtp_peer(), and sip_get_vrtp_peer().

◆ SIP_DIRECT_MEDIA_NAT

#define SIP_DIRECT_MEDIA_NAT (2 << 20)

DP: allow media reinvite when new peer is behind NAT

Definition at line 290 of file sip.h.

Referenced by handle_common_options(), sip_get_rtp_peer(), and sip_set_rtp_peer().

◆ SIP_DTMF

#define SIP_DTMF (7 << 15)

DP: DTMF Support: five settings, uses three bits

Definition at line 275 of file sip.h.

Referenced by __sip_alloc(), _sip_show_peer(), check_peer_ok(), create_addr_from_peer(), dialog_initialize_rtp(), enable_dsp_detect(), handle_common_options(), handle_request_invite(), process_sdp(), sip_dtmfmode(), sip_new(), sip_rtp_read(), sip_senddigit_begin(), sip_senddigit_end(), sip_setoption(), sip_show_channel(), sip_show_settings(), and transmit_info_with_digit().

◆ SIP_DTMF_AUTO

#define SIP_DTMF_AUTO (3 << 15)

DP: DTMF Support: AUTO switch between rfc2833 and in-band DTMF

Definition at line 279 of file sip.h.

Referenced by __sip_alloc(), check_peer_ok(), create_addr_from_peer(), enable_dsp_detect(), handle_common_options(), process_sdp(), sip_dtmfmode(), sip_new(), and sip_setoption().

◆ SIP_DTMF_INBAND

#define SIP_DTMF_INBAND (1 << 15)

DP: DTMF Support: Inband audio, only for ULAW/ALAW - "inband"

Definition at line 277 of file sip.h.

Referenced by enable_dsp_detect(), handle_common_options(), process_sdp(), sip_dtmfmode(), sip_new(), sip_rtp_read(), sip_senddigit_begin(), sip_senddigit_end(), and sip_setoption().

◆ SIP_DTMF_INFO

#define SIP_DTMF_INFO (2 << 15)

DP: DTMF Support: SIP Info messages - "info"

Definition at line 278 of file sip.h.

Referenced by handle_common_options(), sip_dtmfmode(), sip_new(), and sip_senddigit_end().

◆ SIP_DTMF_RFC2833

#define SIP_DTMF_RFC2833 (0 << 15)

DP: DTMF Support: RTP DTMF - "rfc2833"

Definition at line 276 of file sip.h.

Referenced by __sip_alloc(), check_peer_ok(), create_addr_from_peer(), dialog_initialize_rtp(), handle_common_options(), handle_request_invite(), process_sdp(), reload_config(), sip_dtmfmode(), sip_new(), sip_rtp_read(), sip_senddigit_begin(), and sip_senddigit_end().

◆ SIP_DTMF_SHORTINFO

#define SIP_DTMF_SHORTINFO (4 << 15)

DP: DTMF Support: SIP Info messages - "info" - short variant

Definition at line 280 of file sip.h.

Referenced by handle_common_options(), sip_dtmfmode(), sip_new(), sip_senddigit_end(), and transmit_info_with_digit().

◆ SIP_FLAGS_TO_COPY

#define SIP_FLAGS_TO_COPY

Value:

(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
    SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT_FORCE_RPORT | SIP_G726_NONSTANDARD | \
    SIP_USEREQPHONE | SIP_INSECURE | SIP_USEPATH)

Flags to copy from peer/user to dialog.

Definition at line 313 of file sip.h.

Referenced by __sip_alloc(), check_peer_ok(), create_addr_from_peer(), set_peer_defaults(), and sip_poke_peer().

◆ SIP_G726_NONSTANDARD

#define SIP_G726_NONSTANDARD (1 << 31)

DP: Use non-standard packing for G726-32 data

Definition at line 310 of file sip.h.

Referenced by add_codec_to_sdp(), handle_common_options(), and process_sdp_a_audio().

◆ SIP_GOTREFER

#define SIP_GOTREFER (1 << 6)

D: Got a refer?

Definition at line 263 of file sip.h.

Referenced by handle_request_refer(), local_attended_transfer(), and sip_set_rtp_peer().

◆ SIP_INC_COUNT

#define SIP_INC_COUNT (1 << 8)

D: Did this dialog increment the counter of in-use calls?

Definition at line 265 of file sip.h.

Referenced by handle_request_cancel(), sip_hangup(), sip_pvt_dtor(), and update_call_counter().

◆ SIP_INC_RINGING

#define SIP_INC_RINGING (1 << 9)

D: Did this connection increment the counter of in-use calls?

Definition at line 266 of file sip.h.

Referenced by update_call_counter().

◆ SIP_INSECURE

#define SIP_INSECURE (3 << 23)

DP: three settings, uses two bits

Definition at line 294 of file sip.h.

Referenced by _sip_show_peer(), handle_common_options(), load_module(), and sip_reload().

◆ SIP_INSECURE_INVITE

#define SIP_INSECURE_INVITE (1 << 24)

DP: don't require authentication for incoming INVITEs

Definition at line 297 of file sip.h.

Referenced by check_peer_ok(), and set_insecure_flags().

◆ SIP_INSECURE_NONE

#define SIP_INSECURE_NONE (0 << 23)

DP: secure mode

Definition at line 295 of file sip.h.

◆ SIP_INSECURE_PORT

#define SIP_INSECURE_PORT (1 << 23)

DP: don't require matching port for incoming requests

Definition at line 296 of file sip.h.

Referenced by get_insecure_variable_from_config(), get_insecure_variable_from_sipregs(), peer_ipcmp_cb_full(), set_insecure_flags(), and sip_find_peer_full().

◆ SIP_MAX_HEADERS

#define SIP_MAX_HEADERS 64

Max amount of SIP headers to read

Definition at line 111 of file sip.h.

Referenced by add_header(), and parse_request().

◆ SIP_MAX_LINES

#define SIP_MAX_LINES 256

Max amount of lines in SIP attachment (like SDP)

Definition at line 112 of file sip.h.

Referenced by parse_request().

◆ SIP_MAX_PACKET_SIZE

#define SIP_MAX_PACKET_SIZE 20480

Max SIP packet size

Definition at line 113 of file sip.h.

Referenced by sip_tcptls_read().

◆ SIP_MIN_PACKET

#define SIP_MIN_PACKET 4096

Initialize size of memory to allocate for packets

Definition at line 114 of file sip.h.

Referenced by _sip_tcp_helper_thread(), init_req(), init_resp(), and sipsock_read().

◆ SIP_NAT_FORCE_RPORT

#define SIP_NAT_FORCE_RPORT (1 << 18)

DP: Force rport even if not present in the request

Definition at line 283 of file sip.h.

Referenced by _sip_show_peer(), _sip_show_peers_one(), build_peer(), build_via(), check_for_nat(), copy_via_headers(), display_nat_warning(), force_rport_string(), match_nat_options(), parse_register_contact(), peer_ipcmp_cb_full(), register_verify(), reqprep(), send_request(), set_address_from_contact(), set_peer_nat(), sip_nat_mode(), sip_parse_nat_option(), sip_real_dst(), sip_request_call(), sip_show_users(), and transmit_response_using_temp().

◆ SIP_NAT_RPORT_PRESENT

#define SIP_NAT_RPORT_PRESENT (1 << 19)

DP: rport was present in the request

Definition at line 284 of file sip.h.

Referenced by build_via(), check_user_full(), check_via(), copy_via_headers(), parse_register_contact(), and sip_real_dst().

◆ SIP_NEEDREINVITE

#define SIP_NEEDREINVITE (1 << 4)

D: Do we need to send another reinvite?

Definition at line 261 of file sip.h.

Referenced by check_pendings(), interpret_t38_parameters(), sip_hangup(), sip_reinvite_retry(), sip_sendhtml(), sip_set_rtp_peer(), and update_connectedline().

◆ SIP_OFFER_CC

#define SIP_OFFER_CC (1 << 1)

D: Offer CC on subsequent responses

Definition at line 258 of file sip.h.

Referenced by __transmit_response(), sip_cc_agent_init(), and transmit_response_with_sdp().

◆ SIP_OPT_100REL

#define SIP_OPT_100REL (1 << 1)

Definition at line 147 of file sip.h.

◆ SIP_OPT_EARLY_SESSION

#define SIP_OPT_EARLY_SESSION (1 << 3)

Definition at line 149 of file sip.h.

◆ SIP_OPT_EVENTLIST

#define SIP_OPT_EVENTLIST (1 << 11)

Definition at line 157 of file sip.h.

◆ SIP_OPT_FROMCHANGE

#define SIP_OPT_FROMCHANGE (1 << 17)

Definition at line 163 of file sip.h.

◆ SIP_OPT_GRUU

#define SIP_OPT_GRUU (1 << 12)

Definition at line 158 of file sip.h.

◆ SIP_OPT_HISTINFO

#define SIP_OPT_HISTINFO (1 << 15)

Definition at line 161 of file sip.h.

◆ SIP_OPT_JOIN

#define SIP_OPT_JOIN (1 << 4)

Definition at line 150 of file sip.h.

◆ SIP_OPT_NOREFERSUB

#define SIP_OPT_NOREFERSUB (1 << 14)

Definition at line 160 of file sip.h.

◆ SIP_OPT_OUTBOUND

#define SIP_OPT_OUTBOUND (1 << 20)

Definition at line 166 of file sip.h.

◆ SIP_OPT_PATH

#define SIP_OPT_PATH (1 << 5)

Definition at line 151 of file sip.h.

◆ SIP_OPT_PRECONDITION

#define SIP_OPT_PRECONDITION (1 << 7)

Definition at line 153 of file sip.h.

◆ SIP_OPT_PREF

#define SIP_OPT_PREF (1 << 6)

Definition at line 152 of file sip.h.

◆ SIP_OPT_PRIVACY

#define SIP_OPT_PRIVACY (1 << 8)

Definition at line 154 of file sip.h.

◆ SIP_OPT_RECLISTINV

#define SIP_OPT_RECLISTINV (1 << 18)

Definition at line 164 of file sip.h.

◆ SIP_OPT_RECLISTSUB

#define SIP_OPT_RECLISTSUB (1 << 19)

Definition at line 165 of file sip.h.

◆ SIP_OPT_REPLACES

#define SIP_OPT_REPLACES (1 << 0)

Definition at line 146 of file sip.h.

Referenced by AST_TEST_DEFINE().

◆ SIP_OPT_RESPRIORITY

#define SIP_OPT_RESPRIORITY (1 << 16)

Definition at line 162 of file sip.h.

◆ SIP_OPT_SDP_ANAT

#define SIP_OPT_SDP_ANAT (1 << 9)

Definition at line 155 of file sip.h.

◆ SIP_OPT_SEC_AGREE

#define SIP_OPT_SEC_AGREE (1 << 10)

Definition at line 156 of file sip.h.

◆ SIP_OPT_TARGET_DIALOG

#define SIP_OPT_TARGET_DIALOG (1 << 13)

Definition at line 159 of file sip.h.

◆ SIP_OPT_TIMER

#define SIP_OPT_TIMER (1 << 2)

Definition at line 148 of file sip.h.

Referenced by AST_TEST_DEFINE(), handle_request_invite_st(), and respprep().

◆ SIP_OPT_UNKNOWN

#define SIP_OPT_UNKNOWN (1 << 21)

Definition at line 167 of file sip.h.

Referenced by AST_TEST_DEFINE(), and parse_sip_options().

◆ SIP_OUTGOING

#define SIP_OUTGOING (1 << 0)

D: Direction of the last transaction in this dialog

Definition at line 257 of file sip.h.

Referenced by __sip_subscribe_mwi_do(), add_sdp(), handle_request_bye(), handle_request_invite(), handle_request_refer(), handle_response(), handle_response_invite(), manager_sipnotify(), process_crypto(), reqprep(), respprep(), sip_call(), sip_cc_monitor_request_cc(), sip_cli_notify(), sip_hangup(), sip_indicate(), sip_msg_send(), sip_poke_peer(), sip_send_mwi_to_peer(), sip_show_channel(), sip_write(), transmit_publish(), transmit_refer(), transmit_register(), transmit_reinvite_with_sdp(), update_connectedline(), and update_redirecting().

◆ SIP_PAGE2_ALLOWOVERLAP

#define SIP_PAGE2_ALLOWOVERLAP (3 << 13)

DP: Allow overlap dialing ?

Definition at line 337 of file sip.h.

Referenced by _sip_show_peer(), get_destination(), handle_common_options(), handle_request_invite(), handle_response(), sip_indicate(), and sip_show_settings().

◆ SIP_PAGE2_ALLOWOVERLAP_DTMF

#define SIP_PAGE2_ALLOWOVERLAP_DTMF (2 << 13)

Yes, using the DTMF transmission through Early Media

Definition at line 340 of file sip.h.

Referenced by handle_common_options(), and sip_indicate().

◆ SIP_PAGE2_ALLOWOVERLAP_NO

#define SIP_PAGE2_ALLOWOVERLAP_NO (0 << 13)

No, terminate with 404 Not found

Definition at line 338 of file sip.h.

◆ SIP_PAGE2_ALLOWOVERLAP_SPARE

#define SIP_PAGE2_ALLOWOVERLAP_SPARE (3 << 13)

Spare (reserved for another dialling transmission mechanisms like KPML)

Definition at line 341 of file sip.h.

◆ SIP_PAGE2_ALLOWOVERLAP_YES

#define SIP_PAGE2_ALLOWOVERLAP_YES (1 << 13)

Yes, using the 484 Address Incomplete response

Definition at line 339 of file sip.h.

Referenced by handle_common_options(), handle_request_invite(), handle_response(), reload_config(), and sip_indicate().

◆ SIP_PAGE2_ALLOWSUBSCRIBE

#define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 12)

GP: Allow subscriptions from this peer?

Definition at line 335 of file sip.h.

Referenced by _sip_show_peer(), build_peer(), handle_common_options(), handle_request_subscribe(), reload_config(), and sip_show_settings().

◆ SIP_PAGE2_BUGGY_MWI

#define SIP_PAGE2_BUGGY_MWI (1 << 22)

DP: Buggy CISCO MWI fix

Definition at line 357 of file sip.h.

Referenced by handle_common_options(), and transmit_notify_with_mwi().

◆ SIP_PAGE2_CALL_ONHOLD

#define SIP_PAGE2_CALL_ONHOLD (3 << 19)

D: Call hold states:

Definition at line 351 of file sip.h.

Referenced by add_sdp(), change_hold_state(), check_rtp_timeout(), handle_request_cancel(), handle_request_invite(), process_sdp(), show_channels_cb(), sip_hangup(), sip_pvt_dtor(), and update_call_counter().

◆ SIP_PAGE2_CALL_ONHOLD_ACTIVE

#define SIP_PAGE2_CALL_ONHOLD_ACTIVE (1 << 19)

D: Active hold

Definition at line 352 of file sip.h.

Referenced by change_hold_state().

◆ SIP_PAGE2_CALL_ONHOLD_INACTIVE

#define SIP_PAGE2_CALL_ONHOLD_INACTIVE (3 << 19)

D: Inactive hold

Definition at line 354 of file sip.h.

Referenced by add_sdp(), and change_hold_state().

◆ SIP_PAGE2_CALL_ONHOLD_ONEDIR

#define SIP_PAGE2_CALL_ONHOLD_ONEDIR (2 << 19)

D: One directional hold

Definition at line 353 of file sip.h.

Referenced by add_sdp(), and change_hold_state().

◆ SIP_PAGE2_CONNECTLINEUPDATE_PEND

#define SIP_PAGE2_CONNECTLINEUPDATE_PEND (1 << 6)

Definition at line 329 of file sip.h.

Referenced by __transmit_response(), and update_connectedline().

◆ SIP_PAGE2_DIALOG_ESTABLISHED

#define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 23)

29: Has a dialog been established?

Definition at line 358 of file sip.h.

Referenced by check_pendings(), get_sip_pvt_from_replaces(), handle_request_bye(), handle_request_invite(), handle_request_subscribe(), handle_response(), handle_response_invite(), handle_response_message(), match_req_to_dialog(), retrans_pkt(), sip_answer(), sip_cc_agent_destructor(), sip_cc_agent_respond(), and transmit_state_notify().

◆ SIP_PAGE2_FAX_DETECT

#define SIP_PAGE2_FAX_DETECT (3 << 24)

DP: Fax Detection support

Definition at line 360 of file sip.h.

Referenced by handle_common_options(), and reload_config().

◆ SIP_PAGE2_FAX_DETECT_BOTH

#define SIP_PAGE2_FAX_DETECT_BOTH (3 << 24)

DP: Fax Detection support - detect both

Definition at line 363 of file sip.h.

Referenced by handle_common_options().

◆ SIP_PAGE2_FAX_DETECT_CNG

#define SIP_PAGE2_FAX_DETECT_CNG (1 << 24)

DP: Fax Detection support - detect CNG in audio

Definition at line 361 of file sip.h.

Referenced by enable_dsp_detect(), handle_common_options(), and sip_read().

◆ SIP_PAGE2_FAX_DETECT_T38

#define SIP_PAGE2_FAX_DETECT_T38 (2 << 24)

DP: Fax Detection support - detect T.38 reinvite from peer

Definition at line 362 of file sip.h.

Referenced by handle_common_options(), process_sdp(), and sip_call().

◆ SIP_PAGE2_FLAGS_TO_COPY

#define SIP_PAGE2_FLAGS_TO_COPY

Definition at line 375 of file sip.h.

Referenced by __sip_alloc(), check_peer_ok(), create_addr_from_peer(), set_peer_defaults(), and sip_poke_peer().

◆ SIP_PAGE2_HAVEPEERCONTEXT

#define SIP_PAGE2_HAVEPEERCONTEXT (1 << 28) /*< Are we associated with a configured peer context? */

Definition at line 367 of file sip.h.

Referenced by build_peer(), and get_destination().

◆ SIP_PAGE2_IGNORESDPVERSION

#define SIP_PAGE2_IGNORESDPVERSION (1 << 16)

GDP: Ignore the SDP session version number we receive and treat all sessions as new

Definition at line 344 of file sip.h.

Referenced by _sip_show_peer(), handle_common_options(), process_sdp_o(), reload_config(), and sip_show_settings().

◆ SIP_PAGE2_PREFERRED_CODEC

#define SIP_PAGE2_PREFERRED_CODEC (1 << 9)

GDP: Only respond with single most preferred joint codec

Definition at line 332 of file sip.h.

Referenced by build_peer(), process_sdp(), and reload_config().

◆ SIP_PAGE2_Q850_REASON

#define SIP_PAGE2_Q850_REASON (1 << 3)

DP: Get/send cause code via Reason header

Definition at line 326 of file sip.h.

Referenced by __transmit_response(), _sip_show_peer(), build_peer(), reload_config(), sip_show_settings(), transmit_request_with_auth(), and use_reason_header().

◆ SIP_PAGE2_RFC2833_COMPENSATE

#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 21)

DP: Compensate for buggy RFC2833 implementations

Definition at line 356 of file sip.h.

Referenced by create_addr_from_peer(), dialog_initialize_rtp(), handle_common_options(), handle_request_invite(), process_sdp(), and sip_show_settings().

◆ SIP_PAGE2_RPID_IMMEDIATE

#define SIP_PAGE2_RPID_IMMEDIATE (1 << 7)

Definition at line 330 of file sip.h.

Referenced by handle_common_options(), and update_connectedline().

◆ SIP_PAGE2_RPID_UPDATE

#define SIP_PAGE2_RPID_UPDATE (1 << 2)

Definition at line 325 of file sip.h.

Referenced by handle_common_options(), and set_pvt_allowed_methods().

◆ SIP_PAGE2_RPORT_PRESENT

#define SIP_PAGE2_RPORT_PRESENT (1 << 8)

Was rport received in the Via header?

Definition at line 331 of file sip.h.

Referenced by check_user_full(), and check_via().

◆ SIP_PAGE2_RTAUTOCLEAR

#define SIP_PAGE2_RTAUTOCLEAR (1 << 1)

GP: Should we clean memory from peers after expiry?

Definition at line 324 of file sip.h.

Referenced by expire_register(), realtime_peer(), reload_config(), and sip_show_settings().

◆ SIP_PAGE2_RTCACHEFRIENDS

#define SIP_PAGE2_RTCACHEFRIENDS (1 << 0)

GP: Should we keep RT objects in memory for extended time?

Definition at line 323 of file sip.h.

Referenced by build_peer(), parse_register_contact(), realtime_peer(), reload_config(), sip_destroy_peer(), sip_prune_realtime(), sip_show_settings(), and update_peer().

◆ SIP_PAGE2_STATECHANGEQUEUE

#define SIP_PAGE2_STATECHANGEQUEUE (1 << 5)

D: Unsent state pending change exists

Definition at line 328 of file sip.h.

Referenced by extensionstate_update(), and handle_response_notify().

◆ SIP_PAGE2_SUBSCRIBEMWIONLY

#define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 15)

GP: Only issue MWI notification if subscribed to

Definition at line 343 of file sip.h.

Referenced by build_peer(), handle_request_subscribe(), and sip_send_mwi_to_peer().

◆ SIP_PAGE2_SYMMETRICRTP

#define SIP_PAGE2_SYMMETRICRTP (1 << 4)

GDP: Whether symmetric RTP is enabled or not

Definition at line 327 of file sip.h.

Referenced by _sip_show_peer(), _sip_show_peers_one(), check_for_nat(), comedia_string(), do_setnat(), initialize_udptl(), match_nat_options(), process_sdp(), set_peer_nat(), sip_parse_nat_option(), and sip_request_call().

◆ SIP_PAGE2_T38SUPPORT

#define SIP_PAGE2_T38SUPPORT (3 << 17)

GDP: T.38 Fax Support

Definition at line 346 of file sip.h.

Referenced by _sip_show_peer(), check_peer_ok(), handle_t38_options(), initialize_udptl(), interpret_t38_parameters(), set_t38_capabilities(), sip_get_rtp_peer(), sip_queryoption(), and sip_show_settings().

◆ SIP_PAGE2_T38SUPPORT_UDPTL

#define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 17)

GDP: T.38 Fax Support (no error correction)

Definition at line 347 of file sip.h.

Referenced by handle_t38_options(), and set_t38_capabilities().

◆ SIP_PAGE2_T38SUPPORT_UDPTL_FEC

#define SIP_PAGE2_T38SUPPORT_UDPTL_FEC (2 << 17)

GDP: T.38 Fax Support (FEC error correction)

Definition at line 348 of file sip.h.

Referenced by handle_t38_options(), and set_t38_capabilities().

◆ SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY

#define SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY (3 << 17)

GDP: T.38 Fax Support (redundancy error correction)

Definition at line 349 of file sip.h.

Referenced by handle_t38_options(), and set_t38_capabilities().

◆ SIP_PAGE2_TEXTSUPPORT

#define SIP_PAGE2_TEXTSUPPORT (1 << 11)

GDP: Global text enable

Definition at line 334 of file sip.h.

Referenced by _sip_show_peer(), _sip_show_peers_one(), dialog_initialize_rtp(), handle_common_options(), reload_config(), and sip_show_settings().

◆ SIP_PAGE2_TRUST_ID_OUTBOUND

#define SIP_PAGE2_TRUST_ID_OUTBOUND (3 << 30)

DP: Do we trust the peer with private presence information?

Definition at line 370 of file sip.h.

Referenced by _sip_show_peer(), add_rpid(), and handle_common_options().

◆ SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY

#define SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY (0 << 30)

Legacy, Do not provide private presence information, but include PAI/RPID when private

Definition at line 371 of file sip.h.

Referenced by add_rpid(), and handle_common_options().

◆ SIP_PAGE2_TRUST_ID_OUTBOUND_NO

#define SIP_PAGE2_TRUST_ID_OUTBOUND_NO (1 << 30)

No, Do not provide private presence information, do not include PAI/RPID when private

Definition at line 372 of file sip.h.

Referenced by add_rpid(), and handle_common_options().

◆ SIP_PAGE2_TRUST_ID_OUTBOUND_YES

#define SIP_PAGE2_TRUST_ID_OUTBOUND_YES (2 << 30)

Yes, provide private presence information in PAI/RPID headers

Definition at line 373 of file sip.h.

Referenced by add_rpid(), and handle_common_options().

◆ SIP_PAGE2_UDPTL_DESTINATION

#define SIP_PAGE2_UDPTL_DESTINATION (1 << 26)

DP: Use source IP of RTP as destination if NAT is enabled

Definition at line 365 of file sip.h.

Referenced by handle_t38_options(), and process_sdp().

◆ SIP_PAGE2_USE_SRTP

#define SIP_PAGE2_USE_SRTP (1 << 29)

DP: Whether we should offer (only) SRTP

Definition at line 368 of file sip.h.

Referenced by _sip_show_peer(), build_peer(), function_sippeer(), process_sdp(), sip_call(), sip_queryoption(), and sip_setoption().

◆ SIP_PAGE2_VIDEOSUPPORT

#define SIP_PAGE2_VIDEOSUPPORT (1 << 10)

DP: Video supported if offered?

Definition at line 333 of file sip.h.

Referenced by _sip_show_peer(), _sip_show_peers_one(), dialog_initialize_rtp(), handle_common_options(), reload_config(), and sip_show_settings().

◆ SIP_PAGE2_VIDEOSUPPORT_ALWAYS

#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 27)

DP: Always set up video, even if endpoints don't support it

Definition at line 366 of file sip.h.

Referenced by _sip_show_peer(), dialog_initialize_rtp(), handle_common_options(), reload_config(), and sip_new().

◆ SIP_PAGE3_DIRECT_MEDIA_OUTGOING

#define SIP_PAGE3_DIRECT_MEDIA_OUTGOING (1 << 4)

DP: Only send direct media reinvites on outgoing calls

Definition at line 388 of file sip.h.

Referenced by handle_common_options(), and sip_set_rtp_peer().

◆ SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL

#define SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL (1 << 8)

DGP: Stop telling the peer to start music on hold

Definition at line 392 of file sip.h.

Referenced by build_peer(), process_sdp(), and reload_config().

◆ SIP_PAGE3_FLAGS_TO_COPY

#define SIP_PAGE3_FLAGS_TO_COPY

Value:

(SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32 | SIP_PAGE3_NAT_AUTO_RPORT | SIP_PAGE3_NAT_AUTO_COMEDIA | \
    SIP_PAGE3_DIRECT_MEDIA_OUTGOING | SIP_PAGE3_USE_AVPF | SIP_PAGE3_ICE_SUPPORT | SIP_PAGE3_IGNORE_PREFCAPS | \
    SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL | SIP_PAGE3_FORCE_AVP | SIP_PAGE3_RTCP_MUX)

Definition at line 396 of file sip.h.

Referenced by __sip_alloc(), check_peer_ok(), create_addr_from_peer(), set_peer_defaults(), and sip_poke_peer().

◆ SIP_PAGE3_FORCE_AVP

#define SIP_PAGE3_FORCE_AVP (1 << 9)

DGP: Force 'RTP/AVP' for all streams, even DTLS

Definition at line 393 of file sip.h.

Referenced by add_sdp(), and build_peer().

◆ SIP_PAGE3_ICE_SUPPORT

#define SIP_PAGE3_ICE_SUPPORT (1 << 6)

DGP: Enable ICE support

Definition at line 390 of file sip.h.

Referenced by add_sdp(), build_peer(), dialog_initialize_rtp(), and reload_config().

◆ SIP_PAGE3_IGNORE_PREFCAPS

#define SIP_PAGE3_IGNORE_PREFCAPS (1 << 7)

DP: Ignore prefcaps when setting up an outgoing call leg

Definition at line 391 of file sip.h.

Referenced by add_sdp(), and build_peer().

◆ SIP_PAGE3_NAT_AUTO_COMEDIA

#define SIP_PAGE3_NAT_AUTO_COMEDIA (1 << 3)

DGP: Set SIP_PAGE2_SYMMETRICRTP when NAT is detected

Definition at line 387 of file sip.h.

Referenced by _sip_show_peer(), _sip_show_peers_one(), check_for_nat(), comedia_string(), match_nat_options(), set_peer_nat(), sip_parse_nat_option(), and sip_request_call().

◆ SIP_PAGE3_NAT_AUTO_RPORT

#define SIP_PAGE3_NAT_AUTO_RPORT (1 << 2)

DGP: Set SIP_NAT_FORCE_RPORT when NAT is detected

Definition at line 386 of file sip.h.

Referenced by _sip_show_peer(), _sip_show_peers_one(), build_peer(), check_for_nat(), check_peer_ok(), force_rport_string(), match_nat_options(), reload_config(), set_peer_nat(), sip_parse_nat_option(), sip_request_call(), and transmit_response_using_temp().

◆ SIP_PAGE3_RTCP_MUX

#define SIP_PAGE3_RTCP_MUX (1 << 10)

DGP: Attempt to negotiate RFC 5761 RTCP multiplexing

Definition at line 394 of file sip.h.

Referenced by _sip_show_peer(), add_sdp(), configure_rtcp(), handle_common_options(), process_sdp_a_ice(), set_ice_components(), sip_new(), and sip_show_settings().

◆ SIP_PAGE3_SNOM_AOC

#define SIP_PAGE3_SNOM_AOC (1 << 0)

DPG: Allow snom aoc messages

Definition at line 384 of file sip.h.

Referenced by build_peer(), reload_config(), and sip_indicate().

◆ SIP_PAGE3_SRTP_TAG_32

#define SIP_PAGE3_SRTP_TAG_32 (1 << 1)

DP: Use a 32bit auth tag in INVITE not 80bit

Definition at line 385 of file sip.h.

Referenced by add_sdp(), and build_peer().

◆ SIP_PAGE3_USE_AVPF

#define SIP_PAGE3_USE_AVPF (1 << 5)

DGP: Support a minimal AVPF-compatible profile

Definition at line 389 of file sip.h.

Referenced by add_sdp(), build_peer(), and process_sdp().

◆ SIP_PENDINGBYE

#define SIP_PENDINGBYE (1 << 5)

D: Need to send bye after we ack?

Definition at line 262 of file sip.h.

Referenced by check_pendings(), handle_response_invite(), interpret_t38_parameters(), register_verify(), sip_hangup(), sip_sendhtml(), and sip_set_rtp_peer().

◆ SIP_PROG_INBAND

#define SIP_PROG_INBAND (3 << 25)

DP: three settings, uses two bits

Definition at line 300 of file sip.h.

Referenced by handle_common_options(), sip_indicate(), and sip_show_settings().

◆ SIP_PROG_INBAND_NEVER

#define SIP_PROG_INBAND_NEVER (1 << 25)

Definition at line 302 of file sip.h.

Referenced by handle_common_options(), sip_indicate(), and sip_show_settings().

◆ SIP_PROG_INBAND_NO

#define SIP_PROG_INBAND_NO (0 << 25)

Definition at line 301 of file sip.h.

Referenced by sip_show_settings().

◆ SIP_PROG_INBAND_YES

#define SIP_PROG_INBAND_YES (2 << 25)

Definition at line 303 of file sip.h.

Referenced by handle_common_options(), and sip_indicate().

◆ SIP_PROGRESS_SENT

#define SIP_PROGRESS_SENT (1 << 3)

D: Have sent 183 message progress

Definition at line 260 of file sip.h.

Referenced by handle_response_invite(), sip_answer(), sip_indicate(), sip_write(), and update_connectedline().

◆ SIP_PROMISCREDIR

#define SIP_PROMISCREDIR (1 << 11)

DP: Promiscuous redirection

Definition at line 269 of file sip.h.

Referenced by _sip_show_peer(), handle_common_options(), parse_moved_contact(), sip_show_channel(), and sip_show_settings().

◆ sip_ref_peer

#define sip_ref_peer	(	peer,
		tag
	)	ao2_t_bump(peer, tag)

Definition at line 1894 of file sip.h.

Referenced by build_peer(), create_addr(), handle_request_invite(), handle_request_subscribe(), handle_response_peerpoke(), parse_register_contact(), realtime_peer(), reg_source_db(), sip_keepalive_all_peers(), sip_poke_all_peers(), sip_poke_noanswer(), sip_poke_peer(), sip_send_keepalive(), sip_unregister(), and update_call_counter().

◆ SIP_REINVITE

#define SIP_REINVITE (7 << 20)

DP: four settings, uses three bits

Definition at line 287 of file sip.h.

Referenced by handle_common_options(), and sip_call().

◆ SIP_REINVITE_NONE

#define SIP_REINVITE_NONE (0 << 20)

DP: no reinvite allowed

Definition at line 288 of file sip.h.

◆ SIP_REINVITE_UPDATE

#define SIP_REINVITE_UPDATE (4 << 20)

DP: use UPDATE (RFC3311) when reinviting this peer

Definition at line 291 of file sip.h.

Referenced by handle_common_options(), transmit_reinvite_with_sdp(), and update_connectedline().

◆ SIP_RESERVED

#define SIP_RESERVED ";/?:@&=+$,# "

Reserved characters in the username part of the URI

Definition at line 60 of file sip.h.

◆ SIP_RINGING

#define SIP_RINGING (1 << 2)

D: Have sent 180 ringing

Definition at line 259 of file sip.h.

Referenced by sip_indicate(), and update_connectedline().

◆ SIP_SENDRPID

#define SIP_SENDRPID (3 << 29)

DP: Remote Party-ID Support

Definition at line 306 of file sip.h.

Referenced by __transmit_response(), _sip_show_peer(), add_rpid(), handle_common_options(), sip_show_settings(), transmit_invite(), transmit_reinvite_with_sdp(), and update_connectedline().

◆ SIP_SENDRPID_NO

#define SIP_SENDRPID_NO (0 << 29)

Definition at line 307 of file sip.h.

◆ SIP_SENDRPID_PAI

#define SIP_SENDRPID_PAI (1 << 29)

Use "P-Asserted-Identity" for rpid

Definition at line 308 of file sip.h.

Referenced by add_rpid(), and handle_common_options().

◆ SIP_SENDRPID_RPID

#define SIP_SENDRPID_RPID (2 << 29)

Use "Remote-Party-ID" for rpid

Definition at line 309 of file sip.h.

Referenced by handle_common_options().

◆ SIP_TRANS_TIMEOUT

#define SIP_TRANS_TIMEOUT 64 * DEFAULT_TIMER_T1

SIP request timeout (rfc 3261) 64*T1

Todo:: Use known T1 for timeout (peerpoke)

Definition at line 102 of file sip.h.

Referenced by manager_sipnotify(), sip_cli_notify(), and sip_sipredirect().

◆ SIP_TRUSTRPID

#define SIP_TRUSTRPID (1 << 12)

DP: Trust RPID headers?

Definition at line 270 of file sip.h.

Referenced by _sip_show_peer(), get_rpid(), handle_common_options(), and sip_show_settings().

◆ sip_unref_peer

#define sip_unref_peer	(	peer,
		tag
	)	({ ao2_t_cleanup(peer, tag); (NULL); })

Definition at line 1895 of file sip.h.

Referenced by _sip_qualify_peer(), _sip_show_peer(), _sip_show_peers(), _sip_show_peers_one(), build_peer(), check_peer_ok(), complete_sip_peer(), complete_sip_registered_peer(), complete_sip_user(), create_addr(), expire_register(), function_sippeer(), handle_request_invite(), handle_request_notify(), handle_request_subscribe(), handle_response_peerpoke(), manager_sip_peer_status(), match_and_cleanup_peer_sched(), parse_register_contact(), peer_sched_cleanup(), realtime_peer(), receive_message(), reg_source_db(), register_realtime_peers_with_callbackextens(), register_verify(), reload_config(), sip_devicestate(), sip_do_debug_peer(), sip_find_peer_full(), sip_keepalive_all_peers(), sip_msg_send(), sip_poke_all_peers(), sip_poke_noanswer(), sip_poke_peer(), sip_poke_peer_now(), sip_poke_peer_s(), sip_prune_realtime(), sip_pvt_dtor(), sip_report_security_event(), sip_send_keepalive(), sip_show_inuse(), sip_show_user(), sip_show_users(), sip_unregister(), transmit_register(), and update_call_counter().

◆ SIP_USECLIENTCODE

#define SIP_USECLIENTCODE (1 << 14)

DP: Trust X-ClientCode info message

Definition at line 272 of file sip.h.

Referenced by handle_common_options(), handle_request_info(), and sip_show_settings().

◆ SIP_USEPATH

#define SIP_USEPATH (1 << 27)

GDP: Trust and use incoming Path headers?

Definition at line 305 of file sip.h.

Referenced by _sip_show_peer(), add_supported(), build_path(), handle_common_options(), parse_register_contact(), respprep(), and sip_show_settings().

◆ SIP_USEREQPHONE

#define SIP_USEREQPHONE (1 << 13)

DP: Add user=phone to numeric URI. Default off

Definition at line 271 of file sip.h.

Referenced by _sip_show_peer(), build_peer(), initreqprep(), reload_config(), and sip_show_settings().

◆ SIPBUFSIZE

#define SIPBUFSIZE 512

Buffer size for many operations

Definition at line 56 of file sip.h.

Referenced by add_cc_call_info_to_response(), add_diversion(), add_supported(), build_contact(), extract_uri(), initreqprep(), parse_moved_contact(), parse_ok_contact(), parse_register_contact(), process_sdp(), receive_message(), reg_source_db(), respprep(), sip_call(), sip_handle_cc(), transmit_cc_notify(), and transmit_notify_with_sipfrag().

◆ STANDARD_SIP_PORT

#define STANDARD_SIP_PORT 5060

Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS.

Definition at line 176 of file sip.h.

Referenced by __set_address_from_contact(), _sip_show_peer(), ast_sip_ouraddrfor(), AST_TEST_DEFINE(), build_peer(), check_via(), default_sip_port(), initreqprep(), manager_show_registry(), process_via(), reload_config(), set_destination(), sip_parse_host(), sip_parse_register_line(), sip_show_mwi(), sip_show_registry(), sip_show_settings(), sip_standard_port(), transmit_notify_with_mwi(), and transmit_register().

◆ STANDARD_TLS_PORT

#define STANDARD_TLS_PORT 5061

Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS.

Definition at line 178 of file sip.h.

Referenced by __set_address_from_contact(), ast_sip_ouraddrfor(), AST_TEST_DEFINE(), build_peer(), default_sip_port(), reload_config(), set_destination(), sip_parse_host(), sip_parse_register_line(), and sip_standard_port().

◆ SUPPORTED

#define SUPPORTED 1

Define SIP option tags, used in Require: and Supported: headers We need to be aware of these properties in the phones to use the replace: header. We should not do that without knowing that the other end supports it... This is nothing we can configure, we learn by the dialog Supported: header on the REGISTER (peer) or the INVITE (other devices) We are not using many of these today, but will in the future. This is documented in RFC 3261

Definition at line 142 of file sip.h.

Referenced by parse_sip_options().

◆ TRUE

#define TRUE 1

Definition at line 48 of file sip.h.

◆ XMIT_ERROR

#define XMIT_ERROR -2

Definition at line 58 of file sip.h.

Referenced by __sip_reliable_xmit(), __sip_xmit(), handle_response_invite(), retrans_pkt(), sip_call(), sip_poke_peer(), sip_send_keepalive(), and sip_tcptls_write().

Typedef Documentation

◆ esc_publish_callback

typedef int(* const esc_publish_callback) (struct sip_pvt *, struct sip_request *, struct event_state_compositor *, struct sip_esc_entry *)

Definition at line 1761 of file sip.h.

Enumeration Type Documentation

◆ autocreatepeer_mode

enum autocreatepeer_mode

Automatic peer registration behavior.

Enumerator
AUTOPEERS_DISABLED	Automatic peer creation disabled
AUTOPEERS_VOLATILE	Automatic peers dropped on sip reload (pre-1.8 behavior)
AUTOPEERS_PERSIST	Automatic peers survive sip configuration reload

Definition at line 590 of file sip.h.

                        {
    CAN_NOT_CREATE_DIALOG,
    CAN_CREATE_DIALOG,
    CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
 };

◆ can_create_dialog

enum can_create_dialog

States whether a SIP message can create a dialog in Asterisk.

Enumerator
CAN_NOT_CREATE_DIALOG
CAN_CREATE_DIALOG
CAN_CREATE_DIALOG_UNSUPPORTED_METHOD

Definition at line 597 of file sip.h.

605 {

◆ check_auth_result

enum check_auth_result

Authentication result from check_auth* functions.

Enumerator
AUTH_DONT_KNOW	no result, need to check further
AUTH_SUCCESSFUL
AUTH_CHALLENGE_SENT
AUTH_SECRET_FAILED
AUTH_USERNAME_MISMATCH
AUTH_NOT_FOUND	returned by register_verify
AUTH_UNKNOWN_DOMAIN
AUTH_PEER_NOT_DYNAMIC
AUTH_ACL_FAILED
AUTH_BAD_TRANSPORT
AUTH_RTP_FAILED
AUTH_SESSION_LIMIT

Definition at line 517 of file sip.h.

                       {
    REG_STATE_UNREGISTERED = 0,   /*!< We are not registered
        *  \note Initial state. We should have a timeout scheduled for the initial
        * (or next) registration transmission, calling sip_reregister
        */

◆ digest_keys

enum digest_keys

Enumerator
K_RESP
K_URI
K_USER
K_NONCE
K_LAST

Definition at line 702 of file sip.h.

714 {

◆ domain_mode

enum domain_mode

Modes for SIP domain handling in the PBX.

Enumerator
SIP_DOMAIN_AUTO	This domain is auto-configured
SIP_DOMAIN_CONFIG	This domain is from configuration

Definition at line 647 of file sip.h.

651 {

◆ inv_req_result

enum inv_req_result

The results from handling an invite request.

Note: Start at these values so we do not conflict with check_auth_results values when returning from handle_request_invite. check_auth_results only returned during authentication routines

Enumerator
INV_REQ_SUCCESS	Success code
INV_REQ_FAILED	Failure code
INV_REQ_ERROR	Error code

Definition at line 432 of file sip.h.

                   {
    INV_NONE = 0,         /*!< No state at all, maybe not an INVITE dialog */
    INV_CALLING = 1,      /*!< Invite sent, no answer */

◆ invitestates

enum invitestates

States for the INVITE transaction, not the dialog.

Note: this is for the INVITE that sets up the dialog

Enumerator
INV_NONE	No state at all, maybe not an INVITE dialog
INV_CALLING	Invite sent, no answer
INV_PROCEEDING	We got/sent 1xx message
INV_EARLY_MEDIA	We got 18x message with to-tag back
INV_COMPLETED	Got final response with error. Wait for ACK, then CONFIRMED
INV_CONFIRMED	Confirmed response - we've got an ack (Incoming calls only)
INV_TERMINATED	Transaction done - either successful (AST_STATE_UP) or failed, but done The only way out of this is a BYE from one side
INV_CANCELLED	Transaction cancelled by client or server in non-terminated state

Definition at line 441 of file sip.h.

450 {

451 XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.

XMIT_CRITICAL

Definition: sip.h:458

◆ media_type

enum media_type

Media types generate different "dummy answers" for not accepting the offer of a media stream. We need to add definitions for each RTP profile. Secure RTP is not the same as normal RTP and will require a new definition.

Enumerator
SDP_AUDIO	RTP/AVP Audio
SDP_VIDEO	RTP/AVP Video
SDP_IMAGE	Image udptl, not TCP or RTP
SDP_TEXT	RTP/AVP Realtime Text
SDP_UNKNOWN	Unknown media type

Definition at line 489 of file sip.h.

495 {

◆ notifycid_setting

enum notifycid_setting

Settings for the 'notifycid' option, see sip.conf.sample for details.

Enumerator
DISABLED
ENABLED
IGNORE_CONTEXT

Definition at line 640 of file sip.h.

                  {
    SIP_DOMAIN_AUTO,      /*!< This domain is auto-configured */
    SIP_DOMAIN_CONFIG,    /*!< This domain is from configuration */
 };
 

◆ notifyringing_setting

enum notifyringing_setting

Setting for the 'notifyringing' option, see sip.conf.sample for details.

Enumerator
NOTIFYRINGING_DISABLED
NOTIFYRINGING_ENABLED
NOTIFYRINGING_NOTINUSE

Definition at line 633 of file sip.h.

                        {
    DISABLED       = 0,
    ENABLED        = 1,
    IGNORE_CONTEXT = 2,
 };

◆ parse_register_result

enum parse_register_result

Results from the parse_register() function.

Enumerator
PARSE_REGISTER_DENIED
PARSE_REGISTER_FAILED
PARSE_REGISTER_UPDATE
PARSE_REGISTER_QUERY

Definition at line 465 of file sip.h.

                       {
    NONE = 0,
    XPIDF_XML,
    DIALOG_INFO_XML,
    CPIM_PIDF_XML,

◆ referstatus

enum referstatus

Parameters to know status of transfer.

Enumerator
REFER_IDLE	No REFER is in progress
REFER_SENT	Sent REFER to transferee
REFER_RECEIVED	Received REFER from transferrer
REFER_CONFIRMED	Refer confirmed with a 100 TRYING (unused)
REFER_ACCEPTED	Accepted by transferee
REFER_RINGING	Target Ringing
REFER_200OK	Answered by transfer target
REFER_FAILED	REFER declined - go on
REFER_NOAUTH	We had no auth for REFER

Definition at line 674 of file sip.h.

                    {
    SIP_TYPE_PEER = (1 << 0),
    SIP_TYPE_USER = (1 << 1),
 };
 
 enum t38_action_flag {

◆ sip_auth_type

enum sip_auth_type

Authentication types - proxy or www authentication.

Note: Endpoints, like Asterisk, should always use WWW authentication to allow multiple authentications in the same call - to the proxy and to the end point.

Enumerator
PROXY_AUTH
WWW_AUTH

Definition at line 502 of file sip.h.

510 {

◆ sip_cc_notify_state

enum sip_cc_notify_state

The states that can be represented in a SIP call-completion NOTIFY.

Enumerator
CC_QUEUED	Queued, i.e. unavailable
CC_READY	Ready, i.e. available

Definition at line 1552 of file sip.h.

                       {
    /*!
     * \brief Unknown

◆ sip_cc_publish_state

enum sip_cc_publish_state

The states that can be represented in a SIP call-completion PUBLISH.

SIP PUBLISH support! PUBLISH support was added to chan_sip due to its use in the call-completion event package. In order to suspend and unsuspend monitoring of a called party, a PUBLISH message must be sent. Rather than try to hack in PUBLISH transmission and reception solely for the purposes of handling call-completion-related messages, an effort has been made to create a generic framework for handling PUBLISH messages.

There are two main components to the effort, the event publication agent (EPA) and the event state compositor (ESC). Both of these terms appear in RFC 3903, and the implementation in Asterisk conforms to the defintions there. An EPA is a UAC that transmits PUBLISH requests. An ESC is a UAS that receives PUBLISH requests and acts appropriately based on the content of those requests.

ESC: The main structure in chan_sip is the event_state_compositor. There is an event_state_compositor structure for each event package supported (as of Nov 2009 this is only the call-completion package). The structure contains data which is intrinsic to the event package itself, such as the name of the package and a set of callbacks for handling incoming PUBLISH requests. In addition, the event_state_compositor struct contains an ao2_container of sip_esc_entries.

A sip_esc_entry corresponds to an entity which has sent a PUBLISH to Asterisk. We are able to match the incoming PUBLISH to a sip_esc_entry using the Sip-If-Match header of the message. Of course, if none is present, then a new sip_esc_entry will be created.

Once it is determined what type of PUBLISH request has come in (from RFC 3903, it may be an initial, modify, refresh, or remove), then the event package-specific callbacks may be called. If your event package doesn't need to take any specific action for a specific PUBLISH type, it is perfectly safe to not define the callback at all. The callback only needs to take care of application-specific information. If there is a problem, it is up to the callback to take care of sending an appropriate 4xx or 5xx response code. In such a case, the callback should return -1. This will tell the function that called the handler that an appropriate error response has been sent. If the callback returns 0, however, then the caller of the callback will generate a new entity tag and send a 200 OK response.

ESC entries are reference-counted, however as an implementor of a specific event package, this should be transparent, since the reference counts are handled by the general ESC framework.

EPA: The event publication agent in chan_sip is structured quite a bit differently than the ESC. With an ESC, an appropriate entry has to be found based on the contents of an incoming PUBLISH message. With an EPA, the application interested in sending the PUBLISH can maintain a reference to the appropriate EPA entry instead. Similarly, when matching a PUBLISH response to an appropriate EPA entry, the sip_pvt can maintain a reference to the corresponding EPA entry. The result of this train of thought is that there is no compelling reason to maintain a container of these entries.

Instead, there is only the sip_epa_entry structure. Every sip_epa_entry has an entity tag that it maintains so that subsequent PUBLISH requests will be identifiable by the ESC on the far end. In addition, there is a static_data field which contains information that is common to all sip_epa_entries for a specific event package. This static data includes the name of the event package and callbacks for handling specific responses for outgoing PUBLISHes. Also, there is a field for pointing to instance-specific data. This can include the current published state or other identifying information that is specific to an instance of an EPA entry of a particular event package.

When an application wishes to send a PUBLISH request, it simply will call create_epa_entry, followed by transmit_publish in order to send the PUBLISH. That's all that is necessary. Like with ESC entries, sip_epa_entries are reference counted. Unlike ESC entries, though, sip_epa_entries reference counts have to be maintained to some degree by the application making use of the sip_epa_entry. The application will acquire a reference to the EPA entry when it calls create_epa_entry. When the application has finished using the EPA entry (which may not be until after several PUBLISH transactions have taken place) it must use ao2_ref to decrease the reference count by 1.

Enumerator
CC_CLOSED	Closed, i.e. unavailable
CC_OPEN	Open, i.e. available

Definition at line 1542 of file sip.h.

                          {
    /*! Queued, i.e. unavailable */
    CC_QUEUED,

◆ sip_debug_e

enum sip_debug_e

debugging state We store separately the debugging requests from the config file and requests from the CLI. Debugging is enabled if either is set (which means that if sipdebug is set in the config file, we can only turn it off by reloading the config).

Enumerator
sip_debug_none
sip_debug_config
sip_debug_console

Definition at line 658 of file sip.h.

               {
    T38_DISABLED = 0,     /*!< Not enabled */
    T38_LOCAL_REINVITE,   /*!< Offered from local - REINVITE */
    T38_PEER_REINVITE,    /*!< Offered from peer - REINVITE */
    T38_ENABLED,          /*!< Negotiated (enabled) */

◆ sip_get_dest_result

enum sip_get_dest_result

Result from get_destination function.

Enumerator
SIP_GET_DEST_EXTEN_MATCHMORE
SIP_GET_DEST_EXTEN_FOUND
SIP_GET_DEST_EXTEN_NOT_FOUND
SIP_GET_DEST_REFUSED
SIP_GET_DEST_INVALID_URI

Definition at line 508 of file sip.h.

                        {
    AUTH_DONT_KNOW = -100,  /*!< no result, need to check further */
       /* XXX maybe this is the same as AUTH_NOT_FOUND */
    AUTH_SUCCESSFUL = 0,
    AUTH_CHALLENGE_SENT = 1,

◆ sip_mailbox_status

enum sip_mailbox_status

Enumerator
SIP_MAILBOX_STATUS_UNKNOWN
SIP_MAILBOX_STATUS_EXISTING
SIP_MAILBOX_STATUS_NEW

Definition at line 1249 of file sip.h.

1254 {

◆ sip_peer_type

enum sip_peer_type

Enumerator
SIP_TYPE_PEER
SIP_TYPE_USER

Definition at line 686 of file sip.h.

690 {

◆ sip_publish_type

enum sip_publish_type

The types of PUBLISH messages defined in RFC 3903.

Enumerator
SIP_PUBLISH_UNKNOWN	Unknown. This actually is not defined in RFC 3903. We use this as a constant to indicate that an incoming PUBLISH does not fit into any of the other categories and is thus invalid.
SIP_PUBLISH_INITIAL	Initial. The first PUBLISH sent. This will contain a non-zero Expires header as well as a body that indicates the current state of the endpoint that has sent the message. The initial PUBLISH is the only type of PUBLISH to not contain a Sip-If-Match header in it.
SIP_PUBLISH_REFRESH	Refresh. Used to keep a published state from expiring. This will contain a non-zero Expires header but no body since its purpose is not to update state.
SIP_PUBLISH_MODIFY	Modify. Used to change state from its previous value. This will contain a body updating the published state. May or may not contain an Expires header.
SIP_PUBLISH_REMOVE	Remove. Used to remove published state from an ESC. This will contain an Expires header set to 0 and likely no body.
SIP_PUBLISH_UNKNOWN	Unknown. This actually is not defined in RFC 3903. We use this as a constant to indicate that an incoming PUBLISH does not fit into any of the other categories and is thus invalid.
SIP_PUBLISH_INITIAL	Initial. The first PUBLISH sent. This will contain a non-zero Expires header as well as a body that indicates the current state of the endpoint that has sent the message. The initial PUBLISH is the only type of PUBLISH to not contain a Sip-If-Match header in it.
SIP_PUBLISH_REFRESH	Refresh. Used to keep a published state from expiring. This will contain a non-zero Expires header but no body since its purpose is not to update state.
SIP_PUBLISH_MODIFY	Modify. Used to change state from its previous value. This will contain a body updating the published state. May or may not contain an Expires header.
SIP_PUBLISH_REMOVE	Remove. Used to remove published state from an ESC. This will contain an Expires header set to 0 and likely no body.

Definition at line 1562 of file sip.h.

1607 {

1608 /*! The event type */

◆ sip_result

enum sip_result

The result of a lot of functions.

Enumerator
AST_SUCCESS	FALSE means success, funny enough
AST_FAILURE	Failure code

Definition at line 420 of file sip.h.

425 {

◆ sip_tcptls_alert

enum sip_tcptls_alert

Enumerator
TCPTLS_ALERT_DATA	There is new data to be sent out.
TCPTLS_ALERT_STOP	A request to stop the tcp_handler thread.

Definition at line 697 of file sip.h.

714 {

◆ sipmethod

enum sipmethod

SIP Request methods known by Asterisk.

Note: Do NOT make any changes to this enum, or the array following it; if you think you are doing the right thing, you are probably not doing the right thing. If you think there are changes needed, get someone else to review them first before submitting a patch. If these two lists do not match properly bad things will happen.

Enumerator
SIP_UNKNOWN	Unknown response
SIP_RESPONSE	Not request, response to outbound request
SIP_REGISTER	Registration to the mothership, tell us where you are located
SIP_OPTIONS	Check capabilities of a device, used for "ping" too
SIP_NOTIFY	Status update, Part of the event package standard, result of a SUBSCRIBE or a REFER
SIP_INVITE	Set up a session
SIP_ACK	End of a three-way handshake started with INVITE.
SIP_PRACK	Reliable pre-call signalling. Not supported in Asterisk.
SIP_BYE	End of a session
SIP_REFER	Refer to another URI (transfer)
SIP_SUBSCRIBE	Subscribe for updates (voicemail, session status, device status, presence)
SIP_MESSAGE	Text messaging
SIP_UPDATE	Update a dialog. We can send UPDATE; but not accept it
SIP_INFO	Information updates during a session
SIP_CANCEL	Cancel an INVITE
SIP_PUBLISH	Not supported in Asterisk
SIP_PING	Not supported at all, no standard but still implemented out there

Definition at line 612 of file sip.h.

                            {
    NOTIFYRINGING_DISABLED = 0,
    NOTIFYRINGING_ENABLED = 1,
    NOTIFYRINGING_NOTINUSE = 2,
 };

◆ sipregistrystate

enum sipregistrystate

States for outbound registrations (with register= lines in sip.conf.

Enumerator
REG_STATE_UNREGISTERED	We are not registered Note Initial state. We should have a timeout scheduled for the initial (or next) registration transmission, calling sip_reregister
REG_STATE_REGSENT	Registration request sent Note sent initial request, waiting for an ack or a timeout to retransmit the initial request.
REG_STATE_AUTHSENT	We have tried to authenticate Note entered after transmit_register with auth info, waiting for an ack.
REG_STATE_REGISTERED	Registered and done
REG_STATE_REJECTED	Registration rejected Note only used when the remote party has an expire larger than our max-expire. This is a final state from which we do not recover (not sure how correctly).
REG_STATE_TIMEOUT	Registration about to expire, renewing registration
REG_STATE_NOAUTH	We have no accepted credentials Note fatal - no chance to proceed
REG_STATE_FAILED	Registration failed after several tries Note fatal - no chance to proceed

Definition at line 534 of file sip.h.

              {
         SESSION_TIMER_MODE_INVALID = 0, /*!< Invalid value */
         SESSION_TIMER_MODE_ACCEPT,      /*!< Honor inbound Session-Timer requests */
         SESSION_TIMER_MODE_ORIGINATE,   /*!< Originate outbound and honor inbound requests */
         SESSION_TIMER_MODE_REFUSE       /*!< Ignore inbound Session-Timers requests */

◆ st_mode

enum st_mode

Modes in which Asterisk can be configured to run SIP Session-Timers.

Enumerator
SESSION_TIMER_MODE_INVALID	Invalid value
SESSION_TIMER_MODE_ACCEPT	Honor inbound Session-Timer requests
SESSION_TIMER_MODE_ORIGINATE	Originate outbound and honor inbound requests
SESSION_TIMER_MODE_REFUSE	Ignore inbound Session-Timers requests

Definition at line 568 of file sip.h.

                   {
         SESSION_TIMER_REFRESHER_AUTO, /*!< Negotiated                      */
         SESSION_TIMER_REFRESHER_US,   /*!< Initially prefer session refresh by Asterisk */
         SESSION_TIMER_REFRESHER_THEM, /*!< Initially prefer session refresh by the other side */
 };

◆ st_refresher

enum st_refresher

The entity playing the refresher role for Session-Timers.

Enumerator
SESSION_TIMER_REFRESHER_AUTO	Negotiated
SESSION_TIMER_REFRESHER_US	Initially prefer session refresh by Asterisk
SESSION_TIMER_REFRESHER_THEM	Initially prefer session refresh by the other side

Definition at line 576 of file sip.h.

583 {

◆ st_refresher_param

enum st_refresher_param

Enumerator
SESSION_TIMER_REFRESHER_PARAM_UNKNOWN
SESSION_TIMER_REFRESHER_PARAM_UAC
SESSION_TIMER_REFRESHER_PARAM_UAS

Definition at line 582 of file sip.h.

                          {
    AUTOPEERS_DISABLED = 0,    /*!< Automatic peer creation disabled */
    AUTOPEERS_VOLATILE,        /*!< Automatic peers dropped on sip reload (pre-1.8 behavior) */
    AUTOPEERS_PERSIST       /*!< Automatic peers survive sip configuration reload */

◆ subscriptiontype

enum subscriptiontype

Type of subscription, based on the packages we do support, see subscription_types.

Enumerator
NONE
XPIDF_XML
DIALOG_INFO_XML
CPIM_PIDF_XML
PIDF_XML
MWI_NOTIFICATION
CALL_COMPLETION

Definition at line 473 of file sip.h.

482 {

◆ t38_action_flag

enum t38_action_flag

Enumerator
SDP_T38_NONE	Do not modify T38 information at all
SDP_T38_INITIATE	Remote side has requested T38 with us
SDP_T38_ACCEPT	Remote side accepted our T38 request

Definition at line 691 of file sip.h.

695 {

◆ t38state

enum t38state

T38 States for a call.

Enumerator
T38_DISABLED	Not enabled
T38_LOCAL_REINVITE	Offered from local - REINVITE
T38_PEER_REINVITE	Offered from peer - REINVITE
T38_ENABLED	Negotiated (enabled)
T38_REJECTED	Refused

Definition at line 665 of file sip.h.

                  {
    REFER_IDLE,           /*!< No REFER is in progress */
    REFER_SENT,           /*!< Sent REFER to transferee */
    REFER_RECEIVED,       /*!< Received REFER from transferrer */
    REFER_CONFIRMED,      /*!< Refer confirmed with a 100 TRYING (unused) */

◆ transfermodes

enum transfermodes

Authorization scheme for call transfers.

Note: Not a bitfield flag, since there are plans for other modes, like "only allow transfers for authenticated devices"

Enumerator
TRANSFER_OPENFORALL	Allow all SIP transfers
TRANSFER_CLOSED	Allow no SIP transfers

Definition at line 414 of file sip.h.

425 {

◆ xmittype

enum xmittype

When sending a SIP message, we can send with a few options, depending on type of SIP request. UNRELIABLE is moslty used for responses to repeated requests, where the original response would be sent RELIABLE in an INVITE transaction.

Enumerator
XMIT_CRITICAL	Transmit critical SIP message reliably, with re-transmits. If it fails, it's critical and will cause a teardown of the session
XMIT_RELIABLE	Transmit SIP message reliably, with re-transmits
XMIT_UNRELIABLE	Transmit SIP message without bothering with re-transmits

Definition at line 457 of file sip.h.

                            {
    PARSE_REGISTER_DENIED,
    PARSE_REGISTER_FAILED,
    PARSE_REGISTER_UPDATE,
    PARSE_REGISTER_QUERY,

Function Documentation

◆ __init_check_auth_buf()

static void __init_check_auth_buf ( void )

static

Definition at line 1883 of file sip.h.

1888 { ao2_t_cleanup(peer, tag); (NULL); })

ao2_t_cleanup

#define ao2_t_cleanup(obj, tag)

Definition: astobj2.h:1959

NULL

#define NULL

Definition: resample.c:96

◆ sip_auth_headers()

void sip_auth_headers	(	enum sip_auth_type	code,
		char **	header,
		char **	respheader
	)

return the request and response header for a 401 or 407 code

Definition at line 16549 of file chan_sip.c.

References ast_verbose(), PROXY_AUTH, and WWW_AUTH.

Referenced by check_auth(), do_message_auth(), do_proxy_auth(), do_register_auth(), sip_report_security_event(), and transmit_request_with_auth().

 {
    if (code == WWW_AUTH) {       /* 401 */
       *header = "WWW-Authenticate";
       *respheader = "Authorization";
    } else if (code == PROXY_AUTH) { /* 407 */
       *header = "Proxy-Authenticate";
       *respheader = "Proxy-Authorization";
    } else {
       ast_verbose("-- wrong response code %u\n", code);
       *header = *respheader = "Invalid";
    }
 }

◆ sip_find_peer()

struct sip_peer* sip_find_peer	(	const char *	peer,
		struct ast_sockaddr *	addr,
		int	realtime,
		int	which_objects,
		int	devstate_only,
		int	transport
	)

Locate device by name or ip address.

Parameters

peer,addr,realtime,devstate_only,transport
which_objects	Define which objects should be matched when doing a lookup by name. Valid options are FINDUSERS, FINDPEERS, or FINDALLDEVICES. Note that this option is not used at all when doing a lookup by IP.

This is used on find matching device on name or ip/port. If the device was declared as type=peer, we don't match on peer name on incoming INVITEs.

Note: Avoid using this function in new functions if there is a way to avoid it, since it might cause a database lookup.

Definition at line 5851 of file chan_sip.c.

References NULL, and sip_find_peer_full().

Referenced by _sip_qualify_peer(), _sip_show_peer(), check_peer_ok(), create_addr(), function_sippeer(), handle_request_notify(), manager_sip_peer_status(), receive_message(), register_verify(), sip_devicestate(), sip_do_debug_peer(), sip_msg_send(), sip_report_security_event(), sip_show_user(), sip_unregister(), and transmit_register().

 {
    return sip_find_peer_full(peer, addr, NULL, realtime, which_objects, devstate_only, transport);
 }

◆ sip_get_header()

const char* sip_get_header	(	const struct sip_request *	req,
		const char *	name
	)

Get header from SIP request.

Returns: Always return something, so don't check for NULL because it won't happen :-)

Definition at line 8600 of file chan_sip.c.

References __get_header().

Referenced by __find_call(), __sip_alloc(), __transmit_response(), build_route(), cc_handle_publish_error(), change_redirecting_information(), check_auth(), check_user_full(), check_via(), copy_header(), extract_uri(), find_sdp(), get_also_info(), get_destination(), get_pai(), get_rdnis(), get_realm(), get_refer_info(), get_rpid(), gettag(), handle_cc_notify(), handle_cc_subscribe(), handle_incoming(), handle_request_bye(), handle_request_do(), handle_request_info(), handle_request_invite(), handle_request_invite_st(), handle_request_notify(), handle_request_publish(), handle_request_register(), handle_request_subscribe(), handle_request_update(), handle_response(), handle_response_invite(), handle_response_notify(), handle_response_publish(), handle_response_refer(), handle_response_register(), handle_response_subscribe(), handle_response_update(), parse_allowed_methods(), parse_moved_contact(), parse_ok_contact(), parse_oli(), parse_register_contact(), proc_422_rsp(), process_via(), receive_message(), register_verify(), reqprep(), respprep(), send_check_user_failure_response(), send_request(), send_response(), sip_get_cc_information(), sip_pidf_validate(), sip_report_security_event(), sip_sipredirect(), transmit_fake_auth_response(), transmit_invite(), transmit_refer(), transmit_response_with_auth(), transmit_response_with_sdp(), transmit_response_with_t38_sdp(), transmit_state_notify(), uac_sips_contact(), uas_sips_contact(), and use_reason_header().

 {
    int start = 0;
    return __get_header(req, name, &start);
 }

◆ sip_get_transport()

const char* sip_get_transport ( enum ast_transport t )

Return transport as string.

Definition at line 3725 of file chan_sip.c.

References AST_TRANSPORT_TCP, AST_TRANSPORT_TLS, AST_TRANSPORT_UDP, AST_TRANSPORT_WS, and AST_TRANSPORT_WSS.

Referenced by _sip_show_peer(), ast_sip_ouraddrfor(), build_contact(), get_transport_pvt(), handle_request_do(), parse_moved_contact(), sip_report_security_event(), sip_show_settings(), sip_show_tcp(), and transmit_notify_with_mwi().

 {
    switch (t) {
    case AST_TRANSPORT_UDP:
       return "UDP";
    case AST_TRANSPORT_TCP:
       return "TCP";
    case AST_TRANSPORT_TLS:
       return "TLS";
    case AST_TRANSPORT_WS:
    case AST_TRANSPORT_WSS:
       return "WS";
    }
 
    return "UNKNOWN";
 }

Variable Documentation

◆ check_auth_buf

struct ast_threadstorage check_auth_buf = { .once = PTHREAD_ONCE_INIT , .key_init = __init_check_auth_buf , .custom_init = NULL , }

static

Definition at line 1883 of file sip.h.

Referenced by check_auth(), sip_report_security_event(), and transmit_fake_auth_response().

◆ sip_options

const struct cfsip_options sip_options[]

static

Referenced by _sip_show_peer(), add_required_respheader(), parse_sip_options(), and sip_show_channel().

Data Structures

Macros

Enumerations

GlobalSettings

Detailed Description

Macro Definition Documentation

◆ ALLOWED_METHODS

◆ CALLERID_UNKNOWN

◆ CHECK_AUTH_BUF_INITLEN

◆ DEC_CALL_LIMIT

◆ DEC_CALL_RINGING

◆ DEFAULT_ACCEPT_OUTOFCALL_MESSAGE

◆ DEFAULT_ALLOW_EXT_DOM

◆ DEFAULT_ALLOWGUEST

◆ DEFAULT_ALWAYSAUTHREJECT

◆ DEFAULT_AUTH_MESSAGE

◆ DEFAULT_AUTH_OPTIONS

◆ DEFAULT_AUTHLIMIT

◆ DEFAULT_AUTHTIMEOUT

◆ DEFAULT_AUTOCREATEPEER

◆ DEFAULT_CALLCOUNTER

◆ DEFAULT_CALLERID

◆ DEFAULT_COMPACTHEADERS

◆ DEFAULT_CONTEXT

◆ DEFAULT_COS_AUDIO

◆ DEFAULT_COS_SIP

◆ DEFAULT_COS_TEXT

◆ DEFAULT_COS_VIDEO

◆ DEFAULT_DEFAULT_EXPIRY

◆ DEFAULT_DOMAINSASREALM

◆ DEFAULT_ENGINE

◆ DEFAULT_EXPIRY

◆ DEFAULT_FREQ_NOTOK

◆ DEFAULT_KEEPALIVE

◆ DEFAULT_KEEPALIVE_INTERVAL

◆ DEFAULT_LEGACY_USEROPTION_PARSING

◆ DEFAULT_MATCHEXTERNADDRLOCALLY

◆ DEFAULT_MAX_CALL_BITRATE

◆ DEFAULT_MAX_EXPIRY

◆ DEFAULT_MAX_FORWARDS

◆ DEFAULT_MAX_SE

◆ DEFAULT_MAXMS

◆ DEFAULT_MIN_EXPIRY

◆ DEFAULT_MIN_SE

◆ DEFAULT_MOHINTERPRET

◆ DEFAULT_MOHSUGGEST

◆ DEFAULT_MWI_EXPIRY

◆ DEFAULT_MWI_FROM

◆ DEFAULT_NOTIFYCID

◆ DEFAULT_NOTIFYMIME

◆ DEFAULT_NOTIFYRINGING

◆ DEFAULT_PEDANTIC

◆ DEFAULT_QUALIFY

◆ DEFAULT_QUALIFY_GAP

◆ DEFAULT_QUALIFY_PEERS

◆ DEFAULT_QUALIFYFREQ

◆ DEFAULT_REALM

◆ DEFAULT_RECORD_FEATURE

◆ DEFAULT_REGEXTENONQUALIFY

◆ DEFAULT_REGISTRATION_TIMEOUT

◆ DEFAULT_RETRANS

◆ DEFAULT_RTPKEEPALIVE

◆ DEFAULT_SDPOWNER

◆ DEFAULT_SDPSESSION

◆ DEFAULT_SEND_DIVERSION

◆ DEFAULT_SRVLOOKUP

◆ DEFAULT_STORE_SIP_CAUSE

◆ DEFAULT_T1MIN

◆ DEFAULT_TIMER_T1

◆ DEFAULT_TOS_AUDIO

◆ DEFAULT_TOS_SIP

◆ DEFAULT_TOS_TEXT

◆ DEFAULT_TOS_VIDEO

◆ DEFAULT_TRANS_TIMEOUT

◆ DEFAULT_USERAGENT

◆ DEFAULT_VMEXTEN

◆ EXPIRY_GUARD_LIMIT

◆ EXPIRY_GUARD_MIN

◆ EXPIRY_GUARD_PCT

◆ EXPIRY_GUARD_SECS