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Last updated: Sat Feb 3 05:01:12 2007

Asterisk developer's documentation :: Codename Pineapple

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chan_sip.c File Reference

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Detailed Description

Implementation of Session Initiation Protocol.

Author:

Mark Spencer <markster@digium.com>

See Also:

• CREDITS

Implementation of RFC 3261 - without S/MIME, TCP and TLS support Configuration file sip.conf

Todo:

SIP over TCP

SIP over TLS

Better support of forking

VIA branch tag transaction checking

Transaction support

Overview of the handling of SIP sessions

The SIP channel handles several types of SIP sessions, or dialogs, not all of them being "telephone calls".

• Incoming calls that will be sent to the PBX core
• Outgoing calls, generated by the PBX
• SIP subscriptions and notifications of states and voicemail messages
• SIP registrations, both inbound and outbound
• SIP peer management (peerpoke, OPTIONS)
• SIP text messages

In the SIP channel, there's a list of active SIP dialogs, which includes all of these when they are active. "sip show channels" in the CLI will show most of these, excluding subscriptions which are shown by "sip show subscriptions"

incoming packets

Incoming packets are received in the monitoring thread, then handled by sipsock_read(). This function parses the packet and matches an existing dialog or starts a new SIP dialog.

sipsock_read sends the packet to handle_request(), that parses a bit more. if it's a response to an outbound request, it's sent to handle_response(). If it is a request, handle_request sends it to one of a list of functions depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc sipsock_read locks the ast_channel if it exists (an active call) and unlocks it after we have processed the SIP message.

A new INVITE is sent to handle_request_invite(), that will end up starting a new channel in the PBX, the new channel after that executing in a separate channel thread. This is an incoming "call". When the call is answered, either by a bridged channel or the PBX itself the sip_answer() function is called.

The actual media - Video or Audio - is mostly handled by the RTP subsystem in rtp.c

Outbound calls

Outbound calls are set up by the PBX through the sip_request_call() function. After that, they are activated by sip_call().

Hanging up

The PBX issues a hangup on both incoming and outgoing calls through the sip_hangup() function

Definition in file chan_sip.c.

#include "asterisk.h"
#include <stdio.h>
#include <ctype.h>
#include <string.h>
#include <unistd.h>
#include <sys/socket.h>
#include <sys/ioctl.h>
#include <net/if.h>
#include <errno.h>
#include <stdlib.h>
#include <fcntl.h>
#include <netdb.h>
#include <signal.h>
#include <sys/signal.h>
#include <netinet/in.h>
#include <netinet/in_systm.h>
#include <arpa/inet.h>
#include <netinet/ip.h>
#include <regex.h>
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/logger.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/options.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
#include "asterisk/linkedlists.h"
#include "asterisk/stringfields.h"
#include "asterisk/monitor.h"
#include "asterisk/localtime.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/compiler.h"
#include "asterisk/threadstorage.h"
#include "asterisk/translate.h"
#include "asterisk/version.h"

Include dependency graph for chan_sip.c:

Go to the source code of this file.

Data Structures
struct	ast_peer_list
	The peer list: Peers and Friends. More...
struct	ast_register_list
	The register list: Other SIP proxies we register with and place calls to. More...
struct	ast_user_list
	The user list: Users and friends. More...
struct	c_referstatusstring
	Table to convert from REFER status variable to string. More...
struct	cfsip_methods
	Structure for parsing of SIP methods. 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	cfsubscription_types
	Description of SUBSCRIBE events. More...
struct	domain
	Domain data structure. More...
struct	sip_auth
	sip_auth: Credentials for authentication to other SIP services More...
struct	sip_dual
	structure used in transfers 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_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 More...
struct	sip_pvt
	sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe 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_route
	Structure to save routing information for a SIP session. More...
struct	sip_user
	Structure for SIP user data. User's place calls to us. More...
struct	t38properties
	T.38 channel settings (at some point we need to make this alloc'ed. More...
Defines
#define	ALLOWED_METHODS   "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"
	SIP Methods we support.
#define	append_history(p, event, fmt, args...)   append_history_full(p, "%-15s " fmt, event, ## args)
	Append to SIP dialog history.
#define	CALLERID_UNKNOWN   "Unknown"
#define	DEC_CALL_LIMIT   0
#define	DEC_CALL_RINGING   2
#define	DEFAULT_ALLOW_EXT_DOM   TRUE
#define	DEFAULT_ALLOWGUEST   TRUE
#define	DEFAULT_AUTOCREATEPEER   FALSE
#define	DEFAULT_CALLERID   "asterisk"
#define	DEFAULT_COMPACTHEADERS   FALSE
#define	DEFAULT_CONTEXT   "default"
#define	DEFAULT_DEFAULT_EXPIRY   120
#define	DEFAULT_EXPIRY   900
#define	DEFAULT_FREQ_NOTOK   10 * 1000
#define	DEFAULT_FREQ_OK   60 * 1000
#define	DEFAULT_MAX_CALL_BITRATE   (384)
#define	DEFAULT_MAX_EXPIRY   3600
#define	DEFAULT_MAX_FORWARDS   "70"
#define	DEFAULT_MAXMS   2000
#define	DEFAULT_MIN_EXPIRY   60
#define	DEFAULT_MOHINTERPRET   "default"
#define	DEFAULT_MOHSUGGEST   ""
#define	DEFAULT_MWITIME   10
#define	DEFAULT_NOTIFYMIME   "application/simple-message-summary"
#define	DEFAULT_NOTIFYRINGING   TRUE
#define	DEFAULT_PEDANTIC   FALSE
#define	DEFAULT_QUALIFY   FALSE
#define	DEFAULT_REALM   "asterisk"
#define	DEFAULT_REGISTRATION_TIMEOUT   20
#define	DEFAULT_RETRANS   1000
#define	DEFAULT_SRVLOOKUP   FALSE
#define	DEFAULT_T1MIN   100
#define	DEFAULT_TOS_AUDIO   0
#define	DEFAULT_TOS_SIP   0
#define	DEFAULT_TOS_VIDEO   0
#define	DEFAULT_TRANS_TIMEOUT   -1
#define	DEFAULT_USERAGENT   "Asterisk PBX"
#define	DEFAULT_VMEXTEN   "asterisk"
#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	FLAG_FATAL   (1 << 1)
#define	FLAG_RESPONSE   (1 << 0)
#define	FORMAT   "%-15.15s %-10.10s %-11.11s %5.5d/%5.5d %-4.4s %-3.3s %-3.3s %-15.15s %-10.10s\n"
#define	FORMAT   "%-30.30s %-12.12s %8d %-20.20s %-25.25s\n"
#define	FORMAT   "%-40.40s %-20.20s %-16.16s\n"
#define	FORMAT   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %-10s\n"
#define	FORMAT   "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"
#define	FORMAT   "%-25.25s %-15.15s %-15.15s \n"
#define	FORMAT2   "%-15.15s %-10.10s %-11.11s %-11.11s %-4.4s %-7.7s %-15.15s\n"
#define	FORMAT2   "%-30.30s %-12.12s %8.8s %-20.20s %-25.25s\n"
#define	FORMAT2   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %-10s\n"
#define	FORMAT2   "%-25.25s %-15.15s %-15.15s \n"
#define	FORMAT3   "%-15.15s %-10.10s %-11.11s %-15.15s %-13.13s %-15.15s %-10.10s\n"
#define	INC_CALL_LIMIT   1
#define	INC_CALL_RINGING   3
#define	INITIAL_CSEQ   101
#define	IPTOS_MINCOST   0x02
#define	MAX(a, b)   ((a) > (b) ? (a) : (b))
#define	MAX_AUTHTRIES   3
#define	MAX_RETRANS   6
#define	NO_RTP   0
#define	NOT_SUPPORTED   0
#define	RTP   1
#define	SDP_SAMPLE_RATE(x)   (x == AST_FORMAT_G722) ? 16000 : 8000
#define	SIP_ALREADYGONE   (1 << 0)
#define	SIP_CALL_LIMIT   (1 << 28)
#define	SIP_CAN_REINVITE   (1 << 20)
#define	SIP_CAN_REINVITE_NAT   (2 << 20)
#define	SIP_DEFER_BYE_ON_TRANSFER   (1 << 15)
#define	SIP_DTMF   (3 << 16)
#define	SIP_DTMF_AUTO   (3 << 16)
#define	SIP_DTMF_INBAND   (1 << 16)
#define	SIP_DTMF_INFO   (2 << 16)
#define	SIP_DTMF_RFC2833   (0 << 16)
#define	SIP_FLAGS_TO_COPY
#define	SIP_FREE_BIT   (1 << 14)
#define	SIP_G726_NONSTANDARD   (1 << 31)
#define	SIP_GOTREFER   (1 << 7)
#define	SIP_INC_COUNT   (1 << 30)
#define	SIP_INSECURE_INVITE   (1 << 24)
#define	SIP_INSECURE_PORT   (1 << 23)
#define	SIP_MAX_HEADERS   64
#define	SIP_MAX_LINES   64
#define	SIP_MAX_PACKET   4096
#define	SIP_NAT   (3 << 18)
#define	SIP_NAT_ALWAYS   (3 << 18)
#define	SIP_NAT_NEVER   (0 << 18)
#define	SIP_NAT_RFC3581   (1 << 18)
#define	SIP_NAT_ROUTE   (2 << 18)
#define	SIP_NEEDDESTROY   (1 << 1)
#define	SIP_NEEDREINVITE   (1 << 5)
#define	SIP_NO_HISTORY   (1 << 27)
#define	SIP_NOVIDEO   (1 << 2)
#define	SIP_OPT_100REL   (1 << 1)
#define	SIP_OPT_EARLY_SESSION   (1 << 3)
#define	SIP_OPT_EVENTLIST   (1 << 11)
#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_PATH   (1 << 5)
#define	SIP_OPT_PRECONDITION   (1 << 7)
#define	SIP_OPT_PREF   (1 << 6)
#define	SIP_OPT_PRIVACY   (1 << 8)
#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_OUTGOING   (1 << 13)
#define	SIP_PAGE2_ALLOWOVERLAP   (1 << 17)
#define	SIP_PAGE2_ALLOWSUBSCRIBE   (1 << 16)
#define	SIP_PAGE2_BUGGY_MWI   (1 << 26)
#define	SIP_PAGE2_CALL_ONHOLD   (3 << 23)
#define	SIP_PAGE2_CALL_ONHOLD_INACTIVE   (1 << 24)
#define	SIP_PAGE2_CALL_ONHOLD_ONEDIR   (1 << 23)
#define	SIP_PAGE2_DEBUG   (3 << 11)
#define	SIP_PAGE2_DEBUG_CONFIG   (1 << 11)
#define	SIP_PAGE2_DEBUG_CONSOLE   (1 << 12)
#define	SIP_PAGE2_DYNAMIC   (1 << 13)
#define	SIP_PAGE2_FLAGS_TO_COPY
#define	SIP_PAGE2_IGNOREREGEXPIRE   (1 << 10)
#define	SIP_PAGE2_INC_RINGING   (1 << 19)
#define	SIP_PAGE2_RFC2833_COMPENSATE   (1 << 25)
#define	SIP_PAGE2_RT_FROMCONTACT   (1 << 4)
#define	SIP_PAGE2_RTAUTOCLEAR   (1 << 2)
#define	SIP_PAGE2_RTCACHEFRIENDS   (1 << 0)
#define	SIP_PAGE2_RTSAVE_SYSNAME   (1 << 5)
#define	SIP_PAGE2_RTUPDATE   (1 << 1)
#define	SIP_PAGE2_SELFDESTRUCT   (1 << 14)
#define	SIP_PAGE2_SUBSCRIBEMWIONLY   (1 << 18)
#define	SIP_PAGE2_T38SUPPORT   (7 << 20)
#define	SIP_PAGE2_T38SUPPORT_RTP   (2 << 20)
#define	SIP_PAGE2_T38SUPPORT_TCP   (4 << 20)
#define	SIP_PAGE2_T38SUPPORT_UDPTL   (1 << 20)
#define	SIP_PAGE2_VIDEOSUPPORT   (1 << 15)
#define	SIP_PENDINGBYE   (1 << 6)
#define	SIP_PKT_DEBUG   (1 << 0)
#define	SIP_PKT_IGNORE   (1 << 2)
#define	SIP_PKT_WITH_TOTAG   (1 << 1)
#define	SIP_PROG_INBAND   (3 << 25)
#define	SIP_PROG_INBAND_NEVER   (0 << 25)
#define	SIP_PROG_INBAND_NO   (1 << 25)
#define	SIP_PROG_INBAND_YES   (2 << 25)
#define	SIP_PROGRESS_SENT   (1 << 4)
#define	SIP_PROMISCREDIR   (1 << 8)
#define	SIP_REALTIME   (1 << 11)
#define	SIP_REINVITE   (7 << 20)
#define	SIP_REINVITE_UPDATE   (4 << 20)
#define	SIP_RINGING   (1 << 3)
#define	SIP_SENDRPID   (1 << 29)
#define	SIP_TIMER_T1   500
#define	SIP_TRANS_TIMEOUT   32000
#define	SIP_TRUSTRPID   (1 << 9)
#define	SIP_USECLIENTCODE   (1 << 12)
#define	SIP_USEREQPHONE   (1 << 10)
#define	sipdebug   ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)
#define	sipdebug_config   ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)
#define	sipdebug_console   ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)
#define	STANDARD_SIP_PORT   5060
	Standard SIP port from RFC 3261. DO NOT CHANGE THIS.
#define	SUPPORTED   1
#define	SUPPORTED_EXTENSIONS   "replaces"
	SIP Extensions we support.
#define	T38FAX_FILL_BIT_REMOVAL   (1 << 0)
#define	T38FAX_RATE_12000   (1 << 12)
#define	T38FAX_RATE_14400   (1 << 13)
#define	T38FAX_RATE_2400   (1 << 8)
#define	T38FAX_RATE_4800   (1 << 9)
#define	T38FAX_RATE_7200   (1 << 10)
#define	T38FAX_RATE_9600   (1 << 11)
#define	T38FAX_RATE_MANAGEMENT_LOCAL_TCF   (1 << 3)
#define	T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF   (0 << 3)
#define	T38FAX_TRANSCODING_JBIG   (1 << 2)
#define	T38FAX_TRANSCODING_MMR   (1 << 1)
#define	T38FAX_UDP_EC_FEC   (1 << 4)
#define	T38FAX_UDP_EC_NONE   (0 << 4)
#define	T38FAX_UDP_EC_REDUNDANCY   (2 << 4)
#define	T38FAX_VERSION   (3 << 6)
#define	T38FAX_VERSION_0   (0 << 6)
#define	T38FAX_VERSION_1   (1 << 6)
#define	TRUE   1
#define	UNLINK(element, head, prev)
#define	VIDEO_CODEC_MASK   0x1fc0000
Enumerations
enum	can_create_dialog { CAN_NOT_CREATE_DIALOG, CAN_CREATE_DIALOG, CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
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_FAKE_AUTH = -4, AUTH_UNKNOWN_DOMAIN = -5 }
	Authentication result from check_auth* functions. More...
enum	domain_mode { SIP_DOMAIN_AUTO, SIP_DOMAIN_CONFIG }
	Modes for SIP domain handling in the PBX. 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	parse_register_result { PARSE_REGISTER_FAILED, PARSE_REGISTER_UPDATE, PARSE_REGISTER_QUERY }
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_result { AST_SUCCESS = 0, AST_FAILURE = -1 }
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	subscriptiontype {   NONE = 0, XPIDF_XML, DIALOG_INFO_XML, CPIM_PIDF_XML,   PIDF_XML, MWI_NOTIFICATION }
enum	t38state {   T38_DISABLED = 0, T38_LOCAL_DIRECT, T38_LOCAL_REINVITE, T38_PEER_DIRECT,   T38_PEER_REINVITE, T38_ENABLED }
	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 }
Functions
static const char *	__get_header (const struct sip_request *req, const char *name, int *start)
static void	__sip_ack (struct sip_pvt *p, int seqno, int resp, int sipmethod)
	Acknowledges receipt of a packet and stops retransmission.
static int	__sip_autodestruct (void *data)
	Kill a SIP dialog (called by scheduler).
static void	__sip_destroy (struct sip_pvt *p, int lockowner, int lockdialoglist)
	Execute destruction of SIP dialog structure, release memory.
static int	__sip_do_register (struct sip_registry *r)
	Register with SIP proxy.
static void	__sip_pretend_ack (struct sip_pvt *p)
	Pretend to ack all packets maybe the lock on p is not strictly necessary but there might be a race.
static int	__sip_reliable_xmit (struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod)
	Transmit packet with retransmits.
static int	__sip_semi_ack (struct sip_pvt *p, int seqno, int resp, int sipmethod)
	Acks receipt of packet, keep it around (used for provisional responses).
static int	__sip_show_channels (int fd, int argc, char *argv[], int subscriptions)
	SIP show channels CLI (main function).
static int	__sip_xmit (struct sip_pvt *p, char *data, int len)
	Transmit SIP message.
static int	__transmit_response (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
	Base transmit response function.
static int	_sip_show_peer (int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Show one peer in detail (main function).
static int	_sip_show_peers (int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[])
	_sip_show_peers: Execute sip show peers command
static void	add_blank (struct sip_request *req)
	add a blank line if no body
static void	add_codec_to_sdp (const struct sip_pvt *p, int codec, int sample_rate, char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, int debug, int *min_packet_size)
	Add codec offer to SDP offer/answer body in INVITE or 200 OK.
static int	add_digit (struct sip_request *req, char digit, unsigned int duration)
	Add DTMF INFO tone to sip message.
static int	add_header (struct sip_request *req, const char *var, const char *value)
	Add header to SIP message.
static int	add_header_contentLength (struct sip_request *req, int len)
	Add 'Content-Length' header to SIP message.
static int	add_line (struct sip_request *req, const char *line)
	Add content (not header) to SIP message.
static void	add_noncodec_to_sdp (const struct sip_pvt *p, int format, int sample_rate, char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, int debug)
	Add RFC 2833 DTMF offer to SDP.
static struct sip_auth *	add_realm_authentication (struct sip_auth *authlist, char *configuration, int lineno)
	Add realm authentication in list.
static void	add_route (struct sip_request *req, struct sip_route *route)
	Add route header into request per learned route.
static enum sip_result	add_sdp (struct sip_request *resp, struct sip_pvt *p)
	Add Session Description Protocol message.
static int	add_sip_domain (const char *domain, const enum domain_mode mode, const char *context)
	Add SIP domain to list of domains we are responsible for.
static int	add_t38_sdp (struct sip_request *resp, struct sip_pvt *p)
	Add T.38 Session Description Protocol message.
static int	add_text (struct sip_request *req, const char *text)
	Add text body to SIP message.
static struct ast_variable *	add_var (const char *buf, struct ast_variable *list)
static int	add_vidupdate (struct sip_request *req)
	add XML encoded media control with update
static void	append_date (struct sip_request *req)
	Append date to SIP message.
static void	append_history_full (struct sip_pvt *p, const char *fmt,...)
	Append to SIP dialog history with arg list.
static void static void	append_history_va (struct sip_pvt *p, const char *fmt, va_list ap)
	Append to SIP dialog history with arg list.
	AST_LIST_HEAD_NOLOCK (sip_history_head, sip_history)
static	AST_LIST_HEAD_STATIC (domain_list, domain)
	AST_MODULE_INFO (ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT,"Session Initiation Protocol (SIP)",.load=load_module,.unload=unload_module,.reload=reload,)
	AST_MUTEX_DEFINE_STATIC (dialoglock)
	Protect the SIP dialog list (of sip_pvt's).
	AST_MUTEX_DEFINE_STATIC (sip_reload_lock)
	AST_MUTEX_DEFINE_STATIC (monlock)
	Protect the monitoring thread, so only one process can kill or start it, and not when it's doing something critical.
	AST_MUTEX_DEFINE_STATIC (netlock)
static void	ast_quiet_chan (struct ast_channel *chan)
	Turn off generator data XXX Does this function belong in the SIP channel?
static int	ast_sip_ouraddrfor (struct in_addr *them, struct in_addr *us)
	NAT fix - decide which IP address to use for ASterisk server?
	AST_THREADSTORAGE_CUSTOM (ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup)
	A per-thread temporary pvt structure.
static int	attempt_transfer (struct sip_dual *transferer, struct sip_dual *target)
	Attempt transfer of SIP call This fix for attended transfers on a local PBX.
static void	auth_headers (enum sip_auth_type code, char **header, char **respheader)
	return the request and response heade for a 401 or 407 code
static int	auto_congest (void *nothing)
	Scheduled congestion on a call.
static void	build_callid_pvt (struct sip_pvt *pvt)
	Build SIP Call-ID value for a non-REGISTER transaction.
static void	build_callid_registry (struct sip_registry *reg, struct in_addr ourip, const char *fromdomain)
	Build SIP Call-ID value for a REGISTER transaction.
static void	build_contact (struct sip_pvt *p)
	Build contact header - the contact header we send out.
static struct sip_peer *	build_peer (const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime)
	Build peer from configuration (file or realtime static/dynamic).
static int	build_reply_digest (struct sip_pvt *p, int method, char *digest, int digest_len)
	Build reply digest.
static void	build_route (struct sip_pvt *p, struct sip_request *req, int backwards)
	Build route list from Record-Route header.
static void	build_rpid (struct sip_pvt *p)
	Build the Remote Party-ID & From using callingpres options.
static struct sip_user *	build_user (const char *name, struct ast_variable *v, int realtime)
	Initiate a SIP user structure from configuration (configuration or realtime).
static void	build_via (struct sip_pvt *p)
	Build a Via header for a request.
static int	cb_extensionstate (char *context, char *exten, int state, void *data)
	Callback for the devicestate notification (SUBSCRIBE) support subsystem.
static enum check_auth_result	check_auth (struct sip_pvt *p, struct sip_request *req, const char *username, const char *secret, const char *md5secret, int sipmethod, char *uri, enum xmittype reliable, int ignore)
	Check user authorization from peer definition Some actions, like REGISTER and INVITEs from peers require authentication (if peer have secret set).
static enum check_auth_result	check_peer_ok (struct sip_pvt *p, char *of, struct sip_request *req, int sipmethod, struct sockaddr_in *sin, struct sip_peer **authpeer, enum xmittype reliable, char *rpid_num, char *calleridname, char *uri2)
	Validate peer authentication.
static void	check_pendings (struct sip_pvt *p)
	Check pending actions on SIP call.
static void	check_rtp_timeout (struct sip_pvt *dialog, time_t t)
	helper function for the monitoring thread
static int	check_sip_domain (const char *domain, char *context, size_t len)
	check_sip_domain: Check if domain part of uri is local to our server
static int	check_user (struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin)
	Find user If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced.
static enum check_auth_result	check_user_full (struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin, struct sip_peer **authpeer)
	Check if matching user or peer is defined Match user on From: user name and peer on IP/port This is used on first invite (not re-invites) and subscribe requests.
static enum check_auth_result	check_user_ok (struct sip_pvt *p, char *of, struct sip_request *req, int sipmethod, struct sockaddr_in *sin, enum xmittype reliable, char *rpid_num, char *calleridname, char *uri2)
	Validate user authentication.
static void	check_via (struct sip_pvt *p, struct sip_request *req)
	check Via: header for hostname, port and rport request/answer
static void	cleanup_stale_contexts (char *new, char *old)
	Destroy disused contexts between reloads Only used in reload_config so the code for regcontext doesn't get ugly.
static int	clear_realm_authentication (struct sip_auth *authlist)
	Clear realm authentication list (at reload).
static void	clear_sip_domains (void)
	Clear our domain list (at reload).
static char *	complete_sip_debug_peer (const char *line, const char *word, int pos, int state)
	Support routine for 'sip debug peer' CLI.
static char *	complete_sip_peer (const char *word, int state, int flags2)
	Do completion on peer name.
static char *	complete_sip_prune_realtime_peer (const char *line, const char *word, int pos, int state)
	Support routine for 'sip prune realtime peer' CLI.
static char *	complete_sip_prune_realtime_user (const char *line, const char *word, int pos, int state)
	Support routine for 'sip prune realtime user' CLI.
static char *	complete_sip_show_peer (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show peer' CLI.
static char *	complete_sip_show_user (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show user' CLI.
static char *	complete_sip_user (const char *word, int state, int flags2)
	Do completion on user name.
static char *	complete_sipch (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show channel' CLI.
static char *	complete_sipnotify (const char *line, const char *word, int pos, int state)
	Support routine for 'sip notify' CLI.
static int	copy_all_header (struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy all headers from one request to another.
static int	copy_header (struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy one header field from one request to another.
static void	copy_request (struct sip_request *dst, const struct sip_request *src)
	copy SIP request (mostly used to save request for responses)
static struct ast_variable *	copy_vars (struct ast_variable *src)
static int	copy_via_headers (struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy SIP VIA Headers from the request to the response.
static int	create_addr (struct sip_pvt *dialog, const char *opeer)
	create address structure from peer name Or, if peer not found, find it in the global DNS returns TRUE (-1) on failure, FALSE on success
static int	create_addr_from_peer (struct sip_pvt *dialog, struct sip_peer *peer)
	Create address structure from peer reference. return -1 on error, 0 on success.
static void	destroy_association (struct sip_peer *peer)
	Remove registration data from realtime database or AST/DB when registration expires.
static int	determine_firstline_parts (struct sip_request *req)
	Parse first line of incoming SIP request.
static void	dialoglist_lock (void)
	hide the way the list is locked/unlocked
static void	dialoglist_unlock (void)
static void *	do_monitor (void *data)
	The SIP monitoring thread.
static int	do_proxy_auth (struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code, int sipmethod, int init)
	Add authentication on outbound SIP packet.
static int	do_register_auth (struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code)
	Authenticate for outbound registration.
static void	do_setnat (struct sip_pvt *p, int natflags)
	Set nat mode on the various data sockets.
static int	does_peer_need_mwi (struct sip_peer *peer)
	Check whether peer needs a new MWI notification check.
static const char *	domain_mode_to_text (const enum domain_mode mode)
	Print domain mode to cli.
static const char *	dtmfmode2str (int mode)
	Convert DTMF mode to printable string.
static int	expire_register (void *data)
	Expire registration of SIP peer.
static void	extract_uri (struct sip_pvt *p, struct sip_request *req)
	Check Contact: URI of SIP message.
static const char *	find_alias (const char *name, const char *_default)
	Find compressed SIP alias Structure for conversion between compressed SIP and "normal" SIP.
static struct sip_pvt *	find_call (struct sip_request *req, struct sockaddr_in *sin, const int intended_method)
	Connect incoming SIP message to current dialog or create new dialog structure Called by handle_request, sipsock_read.
static const char *	find_closing_quote (const char *start, const char *lim)
	Locate closing quote in a string, skipping escaped quotes. optionally with a limit on the search. start must be past the first quote.
static struct sip_peer *	find_peer (const char *peer, struct sockaddr_in *sin, int realtime)
	Locate peer by name or ip address This is used on incoming SIP message to find matching peer on ip or outgoing message to find matching peer on name.
static struct sip_auth *	find_realm_authentication (struct sip_auth *authlist, const char *realm)
	Find authentication for a specific realm.
static int	find_sdp (struct sip_request *req)
	Determine whether a SIP message contains an SDP in its body.
static int	find_sip_method (const char *msg)
	find_sip_method: Find SIP method from header
static const struct cfsubscription_types *	find_subscription_type (enum subscriptiontype subtype)
	Find subscription type in array.
static struct sip_user *	find_user (const char *name, int realtime)
	Locate user by name Locates user by name (From: sip uri user name part) first from in-memory list (static configuration) then from realtime storage (defined in extconfig.conf).
static void	free_old_route (struct sip_route *route)
	Remove route from route list.
static int	func_check_sipdomain (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	Dial plan function to check if domain is local.
static int	func_header_read (struct ast_channel *chan, const char *function, char *data, char *buf, size_t len)
	Read SIP header (dialplan function).
static int	function_sipchaninfo_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	${SIPCHANINFO()} Dialplan function - reads sip channel data
static int	function_sippeer (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	${SIPPEER()} Dialplan function - reads peer data
static char *	generate_random_string (char *buf, size_t size)
	Generate 32 byte random string for callid's etc.
static int	get_also_info (struct sip_pvt *p, struct sip_request *oreq)
	Call transfer support (old way, deprecated by the IETF)--.
static char *	get_body (struct sip_request *req, char *name)
	Get a specific line from the message body.
static char *	get_body_by_line (const char *line, const char *name, int nameLen)
	Reads one line of SIP message body.
static char *	get_calleridname (const char *input, char *output, size_t outputsize)
	Get caller id name from SIP headers.
static int	get_destination (struct sip_pvt *p, struct sip_request *oreq)
	Find out who the call is for We use the INVITE uri to find out.
static const char *	get_header (const struct sip_request *req, const char *name)
	Get header from SIP request.
static char *	get_in_brackets (char *tmp)
	Pick out text in brackets from character string.
static int	get_msg_text (char *buf, int len, struct sip_request *req)
	Get text out of a SIP MESSAGE packet.
static void	get_our_media_address (struct sip_pvt *p, int needvideo, struct sockaddr_in *sin, struct sockaddr_in *vsin, struct sockaddr_in *dest, struct sockaddr_in *vdest)
	Set all IP media addresses for this call.
static int	get_rdnis (struct sip_pvt *p, struct sip_request *oreq)
	Get referring dnis.
static int	get_refer_info (struct sip_pvt *transferer, struct sip_request *outgoing_req)
	Call transfer support (the REFER method) Extracts Refer headers into pvt dialog structure.
static int	get_rpid_num (const char *input, char *output, int maxlen)
	Get caller id number from Remote-Party-ID header field Returns true if number should be restricted (privacy setting found) output is set to NULL if no number found.
static const char *	get_sdp (struct sip_request *req, const char *name)
	Get a line from an SDP message body.
static const char *	get_sdp_iterate (int *start, struct sip_request *req, const char *name)
	Lookup 'name' in the SDP starting at the 'start' line. Returns the matching line, and 'start' is updated with the next line number.
static struct sip_pvt *	get_sip_pvt_byid_locked (const char *callid, const char *totag, const char *fromtag)
	Lock dialog lock and find matching pvt lock Their tag is fromtag, our tag is to-tag This means that in some transactions, totag needs to be their tag :-) depending upon the direction.
static const char *	gettag (const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize)
	Get tag from packet.
static int	handle_common_options (struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v)
	Handle flag-type options common to configuration of devices - users and peers.
static int	handle_invite_replaces (struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin)
	Handle the transfer part of INVITE with a replaces: header, meaning a target pickup or an attended transfer.
static int	handle_request (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock)
	Handle incoming SIP requests (methods).
static int	handle_request_bye (struct sip_pvt *p, struct sip_request *req)
	Handle incoming BYE request.
static int	handle_request_cancel (struct sip_pvt *p, struct sip_request *req)
	Handle incoming CANCEL request.
static void	handle_request_info (struct sip_pvt *p, struct sip_request *req)
	Receive SIP INFO Message.
static int	handle_request_invite (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *recount, char *e)
	Handle incoming INVITE request.
static int	handle_request_message (struct sip_pvt *p, struct sip_request *req)
	Handle incoming MESSAGE request.
static int	handle_request_notify (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e)
	Handle incoming notifications.
static int	handle_request_options (struct sip_pvt *p, struct sip_request *req)
	Handle incoming OPTIONS request.
static int	handle_request_refer (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, int *nounlock)
static int	handle_request_register (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e)
	Handle incoming REGISTER request.
static int	handle_request_subscribe (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e)
	Handle incoming SUBSCRIBE request.
static void	handle_response (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
	Handle SIP response in dialogue.
static void	handle_response_invite (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
	Handle SIP response to INVITE dialogue.
static void	handle_response_peerpoke (struct sip_pvt *p, int resp, struct sip_request *req)
	Handle qualification responses (OPTIONS).
static void	handle_response_refer (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
static int	handle_response_register (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
	Handle responses on REGISTER to services.
static const char *	hangup_cause2sip (int cause)
	Convert Asterisk hangup causes to SIP codes.
static int	hangup_sip2cause (int cause)
	Convert SIP hangup causes to Asterisk hangup causes.
static int	init_req (struct sip_request *req, int sipmethod, const char *recip)
	Initialize SIP request.
static int	init_resp (struct sip_request *resp, const char *msg)
	Initialize SIP response, based on SIP request.
static void	initialize_initreq (struct sip_pvt *p, struct sip_request *req)
	Initialize the initital request packet in the pvt structure. This packet is used for creating replies and future requests in a dialog.
static void	initreqprep (struct sip_request *req, struct sip_pvt *p, int sipmethod)
	Initiate new SIP request to peer/user.
static const char *	insecure2str (int port, int invite)
	Convert Insecure setting to printable string.
static void	list_route (struct sip_route *route)
	List all routes - mostly for debugging.
static int	load_module (void)
	PBX load module - initialization.
static int	local_attended_transfer (struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno)
	Find all call legs and bridge transferee with target called from handle_request_refer.
static int	lws2sws (char *msgbuf, int len)
	Parse multiline SIP headers into one header This is enabled if pedanticsipchecking is enabled.
static void	make_our_tag (char *tagbuf, size_t len)
	Make our SIP dialog tag.
static int	manager_sip_show_peer (struct mansession *s, const struct message *m)
	Show SIP peers in the manager API.
static int	manager_sip_show_peers (struct mansession *s, const struct message *m)
	Show SIP peers in the manager API.
static int	method_match (enum sipmethod id, const char *name)
	returns true if 'name' (with optional trailing whitespace) matches the sip method 'id'. Strictly speaking, SIP methods are case SENSITIVE, but we do a case-insensitive comparison to be more tolerant. following Jon Postel's rule: Be gentle in what you accept, strict with what you send
static char *	nat2str (int nat)
	Convert NAT setting to text string.
static void	parse_copy (struct sip_request *dst, const struct sip_request *src)
	Copy SIP request, parse it.
static void	parse_moved_contact (struct sip_pvt *p, struct sip_request *req)
	Parse 302 Moved temporalily response.
static int	parse_ok_contact (struct sip_pvt *pvt, struct sip_request *req)
	Save contact header for 200 OK on INVITE.
static enum parse_register_result	parse_register_contact (struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
	Parse contact header and save registration (peer registration).
static void	parse_request (struct sip_request *req)
	Parse a SIP message.
static unsigned int	parse_sip_options (struct sip_pvt *pvt, const char *supported)
	Parse supported header in incoming packet.
static int	parse_uri (char *uri, char *scheme, char **ret_name, char **pass, char **domain, char **port, char **options)
static int	peer_status (struct sip_peer *peer, char *status, int statuslen)
	Report Peer status in character string.
static void	print_codec_to_cli (int fd, struct ast_codec_pref *pref)
	Print codec list from preference to CLI/manager.
static void	print_group (int fd, ast_group_t group, int crlf)
	Print call group and pickup group.
static int	process_sdp (struct sip_pvt *p, struct sip_request *req)
	Process SIP SDP offer, select formats and activate RTP channels If offer is rejected, we will not change any properties of the call Return 0 on success, a negative value on errors. Must be called after find_sdp().
static struct sip_peer *	realtime_peer (const char *newpeername, struct sockaddr_in *sin)
	realtime_peer: Get peer from realtime storage Checks the "sippeers" realtime family from extconfig.conf
static void	realtime_update_peer (const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey)
	Update peer object in realtime storage If the Asterisk system name is set in asterisk.conf, we will use that name and store that in the "regserver" field in the sippeers table to facilitate multi-server setups.
static struct sip_user *	realtime_user (const char *username)
	Load user from realtime storage Loads user from "sipusers" category in realtime (extconfig.conf) Users are matched on From: user name (the domain in skipped).
static void	receive_message (struct sip_pvt *p, struct sip_request *req)
	Receive SIP MESSAGE method messages.
static const char *	referstatus2str (enum referstatus rstatus)
	Convert transfer status to string.
static void	reg_source_db (struct sip_peer *peer)
	Get registration details from Asterisk DB.
static void	register_peer_exten (struct sip_peer *peer, int onoff)
	Automatically add peer extension to dial plan.
static enum check_auth_result	register_verify (struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req, char *uri)
	Verify registration of user Registration is done in several steps, first a REGISTER without auth to get a challenge (nonce) then a second one with auth Registration requests are only matched with peers that are marked as "dynamic".
static struct sip_registry *	registry_addref (struct sip_registry *reg)
	Add object reference to SIP registry.
static void	registry_unref (struct sip_registry *reg)
static char *	regstate2str (enum sipregistrystate regstate)
	Convert registration state status to string.
static int	reload (void)
	Part of Asterisk module interface.
static int	reload_config (enum channelreloadreason reason)
	Re-read SIP.conf config file.
static void	replace_cid (struct sip_pvt *p, const char *rpid_num, const char *calleridname)
	helper function for check_{user|peer}_ok()
static int	reply_digest (struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len)
	reply to authentication for outbound registrations
static int	reqprep (struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch)
	Initialize a SIP request message (not the initial one in a dialog).
static int	respprep (struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Prepare SIP response packet.
static int	restart_monitor (void)
	Start the channel monitor thread.
static int	retrans_pkt (void *data)
	Retransmit SIP message if no answer (Called from scheduler).
static int	send_request (struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
	Send SIP Request to the other part of the dialogue.
static int	send_response (struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
	Transmit response on SIP request.
static int	set_address_from_contact (struct sip_pvt *pvt)
	Change the other partys IP address based on given contact.
static void	set_destination (struct sip_pvt *p, char *uri)
	Set destination from SIP URI.
static void	set_peer_defaults (struct sip_peer *peer)
	Set peer defaults before configuring specific configurations.
static int	sip_addheader (struct ast_channel *chan, void *data)
	Add a SIP header to an outbound INVITE.
static int	sip_addrcmp (char *name, struct sockaddr_in *sin)
	Support routine for find_peer.
static struct sip_pvt *	sip_alloc (ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method)
	Allocate SIP_PVT structure and set defaults.
static void	sip_alreadygone (struct sip_pvt *dialog)
	Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging.
static int	sip_answer (struct ast_channel *ast)
	sip_answer: Answer SIP call , send 200 OK on Invite Part of PBX interface
static int	sip_call (struct ast_channel *ast, char *dest, int timeout)
	Initiate SIP call from PBX used from the dial() application.
static void	sip_cancel_destroy (struct sip_pvt *p)
	Cancel destruction of SIP dialog.
static int	sip_debug_test_addr (const struct sockaddr_in *addr)
	See if we pass debug IP filter.
static int	sip_debug_test_pvt (struct sip_pvt *p)
	Test PVT for debugging output.
static void	sip_destroy (struct sip_pvt *p)
	Destroy SIP call structure.
static void	sip_destroy_peer (struct sip_peer *peer)
	Destroy peer object from memory.
static void	sip_destroy_user (struct sip_user *user)
	Remove user object from in-memory storage.
static int	sip_devicestate (void *data)
	Part of PBX channel interface.
static int	sip_do_debug (int fd, int argc, char *argv[])
	Turn on SIP debugging (CLI command).
static int	sip_do_debug_ip (int fd, int argc, char *argv[])
	Enable SIP Debugging in CLI.
static int	sip_do_debug_peer (int fd, int argc, char *argv[])
	sip_do_debug_peer: Turn on SIP debugging with peer mask
static int	sip_do_history (int fd, int argc, char *argv[])
	Enable SIP History logging (CLI).
static int	sip_do_reload (enum channelreloadreason reason)
	Reload module.
static int	sip_dtmfmode (struct ast_channel *chan, void *data)
	Set the DTMFmode for an outbound SIP call (application).
static void	sip_dump_history (struct sip_pvt *dialog)
	Dump SIP history to debug log file at end of lifespan for SIP dialog.
static int	sip_fixup (struct ast_channel *oldchan, struct ast_channel *newchan)
	sip_fixup: Fix up a channel: If a channel is consumed, this is called. Basically update any ->owner links
static int	sip_get_codec (struct ast_channel *chan)
	Return SIP UA's codec (part of the RTP interface).
static enum ast_rtp_get_result	sip_get_rtp_peer (struct ast_channel *chan, struct ast_rtp **rtp)
	Returns null if we can't reinvite audio (part of RTP interface).
static struct ast_udptl *	sip_get_udptl_peer (struct ast_channel *chan)
static enum ast_rtp_get_result	sip_get_vrtp_peer (struct ast_channel *chan, struct ast_rtp **rtp)
	Returns null if we can't reinvite video (part of RTP interface).
static int	sip_handle_t38_reinvite (struct ast_channel *chan, struct sip_pvt *pvt, int reinvite)
	Handle T38 reinvite.
static int	sip_hangup (struct ast_channel *ast)
	sip_hangup: Hangup SIP call Part of PBX interface, called from ast_hangup
static int	sip_indicate (struct ast_channel *ast, int condition, const void *data, size_t datalen)
	Play indication to user With SIP a lot of indications is sent as messages, letting the device play the indication - busy signal, congestion etc.
static const char *	sip_nat_mode (const struct sip_pvt *p)
	Display SIP nat mode.
static struct ast_channel *	sip_new (struct sip_pvt *i, int state, const char *title)
	Initiate a call in the SIP channel called from sip_request_call (calls from the pbx ) for outbound channels and from handle_request_invite for inbound channels.
static int	sip_no_debug (int fd, int argc, char *argv[])
	Disable SIP Debugging in CLI.
static int	sip_no_history (int fd, int argc, char *argv[])
	Disable SIP History logging (CLI).
static int	sip_notify (int fd, int argc, char *argv[])
	Cli command to send SIP notify to peer.
static int	sip_park (struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno)
	Park a call using the subsystem in res_features.c This is executed in a separate thread.
static void *	sip_park_thread (void *stuff)
	Park SIP call support function Starts in a new thread, then parks the call XXX Should we add a wait period after streaming audio and before hangup?? Sometimes the audio can't be heard before hangup.
static void	sip_peer_hold (struct sip_pvt *p, int hold)
	Change onhold state of a peer using a pvt structure.
static void	sip_poke_all_peers (void)
	Send a poke to all known peers Space them out 100 ms apart XXX We might have a cool algorithm for this or use random - any suggestions?
static int	sip_poke_noanswer (void *data)
	React to lack of answer to Qualify poke.
static int	sip_poke_peer (struct sip_peer *peer)
	Check availability of peer, also keep NAT open.
static int	sip_poke_peer_s (void *data)
	Poke peer (send qualify to check if peer is alive and well).
static int	sip_prune_realtime (int fd, int argc, char *argv[])
	Remove temporary realtime objects from memory (CLI).
static void	sip_pvt_lock (struct sip_pvt *pvt)
	Helper function to lock, hiding the underlying locking mechanism.
static void	sip_pvt_unlock (struct sip_pvt *pvt)
	Helper function to unlock pvt, hiding the underlying locking mechanism.
static struct ast_frame *	sip_read (struct ast_channel *ast)
	Read SIP RTP from channel.
static const struct sockaddr_in *	sip_real_dst (const struct sip_pvt *p)
	The real destination address for a write.
static int	sip_refer_allocate (struct sip_pvt *p)
	Allocate SIP refer structure.
static int	sip_reg_timeout (void *data)
	Registration timeout, register again.
static int	sip_register (char *value, int lineno)
	Parse register=> line in sip.conf and add to registry.
static void	sip_registry_destroy (struct sip_registry *reg)
	Destroy registry object Objects created with the register= statement in static configuration.
static int	sip_reload (int fd, int argc, char *argv[])
	Force reload of module from cli.
static struct ast_channel *	sip_request_call (const char *type, int format, void *data, int *cause)
	PBX interface function -build SIP pvt structure SIP calls initiated by the PBX arrive here.
static int	sip_reregister (void *data)
	Update registration with SIP Proxy.
static struct ast_frame *	sip_rtp_read (struct ast_channel *ast, struct sip_pvt *p, int *faxdetect)
	Read RTP from network.
static void	sip_scheddestroy (struct sip_pvt *p, int ms)
	Schedule destruction of SIP dialog.
static void	sip_send_all_registers (void)
	Send all known registrations.
static int	sip_send_mwi_to_peer (struct sip_peer *peer)
	Send message waiting indication to alert peer that they've got voicemail.
static int	sip_senddigit_begin (struct ast_channel *ast, char digit)
static int	sip_senddigit_end (struct ast_channel *ast, char digit, unsigned int duration)
	Send DTMF character on SIP channel within one call, we're able to transmit in many methods simultaneously.
static int	sip_sendtext (struct ast_channel *ast, const char *text)
	Send SIP MESSAGE text within a call Called from PBX core sendtext() application.
static void	sip_set_redirstr (struct sip_pvt *p, char *reason)
	Translate referring cause.
static int	sip_set_rtp_peer (struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
	Set the RTP peer for this call.
static int	sip_set_udptl_peer (struct ast_channel *chan, struct ast_udptl *udptl)
static int	sip_show_channel (int fd, int argc, char *argv[])
	Show details of one active dialog.
static int	sip_show_channels (int fd, int argc, char *argv[])
	Show active SIP channels.
static int	sip_show_domains (int fd, int argc, char *argv[])
	CLI command to list local domains.
static int	sip_show_history (int fd, int argc, char *argv[])
	Show history details of one dialog.
static int	sip_show_inuse (int fd, int argc, char *argv[])
	CLI Command to show calls within limits set by call_limit.
static int	sip_show_objects (int fd, int argc, char *argv[])
	List all allocated SIP Objects (realtime or static).
static int	sip_show_peer (int fd, int argc, char *argv[])
	Show one peer in detail.
static int	sip_show_peers (int fd, int argc, char *argv[])
	CLI Show Peers command.
static int	sip_show_registry (int fd, int argc, char *argv[])
	Show SIP Registry (registrations with other SIP proxies.
static int	sip_show_settings (int fd, int argc, char *argv[])
	List global settings for the SIP channel.
static int	sip_show_subscriptions (int fd, int argc, char *argv[])
	Show active SIP subscriptions.
static int	sip_show_user (int fd, int argc, char *argv[])
	Show one user in detail.
static int	sip_show_users (int fd, int argc, char *argv[])
	CLI Command 'SIP Show Users'.
static int	sip_sipredirect (struct sip_pvt *p, const char *dest)
	Transfer call before connect with a 302 redirect.
static int	sip_transfer (struct ast_channel *ast, const char *dest)
	Transfer SIP call.
static int	sip_write (struct ast_channel *ast, struct ast_frame *frame)
	Send frame to media channel (rtp).
static int	sipsock_read (int *id, int fd, short events, void *ignore)
	Read data from SIP socket.
static void	stop_media_flows (struct sip_pvt *p)
	Immediately stop RTP, VRTP and UDPTL as applicable.
static const char *	subscription_type2str (enum subscriptiontype subtype)
	Show subscription type in string format.
static int	t38_get_rate (int t38cap)
	Get Max T.38 Transmission rate from T38 capabilities.
static struct sip_peer *	temp_peer (const char *name)
	Create temporary peer (used in autocreatepeer mode).
static void	temp_pvt_cleanup (void *)
static int	temp_pvt_init (void *)
static char *	terminate_uri (char *uri)
static char *	transfermode2str (enum transfermodes mode)
	Convert transfer mode to text string.
static void	transmit_fake_auth_response (struct sip_pvt *p, struct sip_request *req, int reliable)
	Send a fake 401 Unauthorized response when the administrator wants to hide the names of local users/peers from fishers.
static int	transmit_info_with_digit (struct sip_pvt *p, const char digit, unsigned int duration)
	Send SIP INFO dtmf message, see Cisco documentation on cisco.com.
static int	transmit_info_with_vidupdate (struct sip_pvt *p)
	Send SIP INFO with video update request.
static int	transmit_invite (struct sip_pvt *p, int sipmethod, int sdp, int init)
	Build REFER/INVITE/OPTIONS message and transmit it.
static int	transmit_message_with_text (struct sip_pvt *p, const char *text)
	Transmit text with SIP MESSAGE method.
static int	transmit_notify_with_mwi (struct sip_pvt *p, int newmsgs, int oldmsgs, char *vmexten)
	Notify user of messages waiting in voicemail.
static int	transmit_notify_with_sipfrag (struct sip_pvt *p, int cseq, char *message, int terminate)
	Notify a transferring party of the status of transfer.
static int	transmit_refer (struct sip_pvt *p, const char *dest)
	Transmit SIP REFER message (initiated by the transfer() dialplan application.
static int	transmit_register (struct sip_registry *r, int sipmethod, const char *auth, const char *authheader)
	Transmit register to SIP proxy or UA.
static int	transmit_reinvite_with_sdp (struct sip_pvt *p, int t38version)
	Transmit reinvite with SDP.
static int	transmit_request (struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
	Transmit generic SIP request.
static int	transmit_request_with_auth (struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
	Transmit SIP request, auth added.
static int	transmit_response (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Transmit response, no retransmits.
static int	transmit_response_reliable (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Transmit response, Make sure you get an ACK This is only used for responses to INVITEs, where we need to make sure we get an ACK.
static int	transmit_response_using_temp (ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg)
	Transmit response, no retransmits, using a temporary pvt structure.
static int	transmit_response_with_allow (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
	Append Accept header, content length before transmitting response.
static int	transmit_response_with_auth (struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *randdata, enum xmittype reliable, const char *header, int stale)
	Respond with authorization request.
static int	transmit_response_with_date (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Append date and content length before transmitting response.
static int	transmit_response_with_sdp (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
	Used for 200 OK and 183 early media.
static int	transmit_response_with_t38_sdp (struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
	Used for 200 OK and 183 early media.
static int	transmit_response_with_unsupported (struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported)
	Transmit response, no retransmits.
static int	transmit_sip_request (struct sip_pvt *p, struct sip_request *req)
	Transmit SIP request unreliably (only used in sip_notify subsystem).
static int	transmit_state_notify (struct sip_pvt *p, int state, int full, int timeout)
	Used in the SUBSCRIBE notification subsystem.
static void	try_suggested_sip_codec (struct sip_pvt *p)
	Try setting codec suggested by the SIP_CODEC channel variable.
static int	unload_module (void)
	PBX unload module API.
static void	unref_peer (struct sip_peer *peer)
static void	unref_user (struct sip_user *user)
static int	update_call_counter (struct sip_pvt *fup, int event)
	update_call_counter: Handle call_limit for SIP users Setting a call-limit will cause calls above the limit not to be accepted.
static void	update_peer (struct sip_peer *p, int expiry)
	Update peer data in database (if used).
Variables
static struct in_addr	__ourip
static int	allow_external_domains
static int	apeerobjs = 0
static char *	app_dtmfmode = "SIPDtmfMode"
static char *	app_sipaddheader = "SIPAddHeader"
static struct sip_auth *	authl = NULL
static int	autocreatepeer
static struct sockaddr_in	bindaddr = { 0, }
static struct ast_custom_function	checksipdomain_function
static struct ast_cli_entry	cli_sip []
	SIP Cli commands definition.
static int	compactheaders
static const char	config [] = "sip.conf"
static const char	debug_usage []
static struct sockaddr_in	debugaddr
static char	default_callerid [AST_MAX_EXTENSION]
static char	default_context [AST_MAX_CONTEXT]
static int	default_expiry = DEFAULT_DEFAULT_EXPIRY
static char	default_fromdomain [AST_MAX_EXTENSION]
static struct ast_jb_conf	default_jbconf
	Global jitterbuffer configuration - by default, jb is disabled.
static char	default_language [MAX_LANGUAGE]
static int	default_maxcallbitrate
static char	default_mohinterpret [MAX_MUSICCLASS]
static char	default_mohsuggest [MAX_MUSICCLASS]
static char	default_notifymime [AST_MAX_EXTENSION]
static struct ast_codec_pref	default_prefs
static int	default_qualify
static char	default_subscribecontext [AST_MAX_CONTEXT]
static char	default_vmexten [AST_MAX_EXTENSION]
static char *	descrip_dtmfmode = "SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call\n"
static char *	descrip_sipaddheader
static struct sip_pvt *	dialoglist = NULL
static int	dumphistory
static int	expiry = DEFAULT_EXPIRY
static time_t	externexpire = 0
static char	externhost [MAXHOSTNAMELEN]
static struct sockaddr_in	externip
static int	externrefresh = 10
static int	global_allowguest
static int	global_allowsubscribe
static enum transfermodes	global_allowtransfer
static int	global_alwaysauthreject
static int	global_autoframing
static int	global_callevents
static int	global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263
	Codecs that we support by default:.
static int	global_directrtpsetup
static struct ast_flags	global_flags [2] = {{0}}
static struct ast_jb_conf	global_jbconf
static int	global_limitonpeers
static int	global_match_auth_username
static int	global_mwitime
static int	global_notifyhold
static int	global_notifyringing
static char	global_realm [MAXHOSTNAMELEN]
static int	global_reg_timeout
static int	global_regattempts_max
static char	global_regcontext [AST_MAX_CONTEXT]
static int	global_relaxdtmf
static int	global_rtautoclear
static int	global_rtpholdtimeout
static int	global_rtpkeepalive
static int	global_rtptimeout
static int	global_srvlookup
static int	global_t1min
static int	global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600
static unsigned int	global_tos_audio
static unsigned int	global_tos_sip
static unsigned int	global_tos_video
static char	global_useragent [AST_MAX_EXTENSION]
static const char	history_usage []
static struct io_context *	io
static struct ast_ha *	localaddr
static char	mandescr_show_peer []
static char	mandescr_show_peers []
static int	max_expiry = DEFAULT_MAX_EXPIRY
static int	min_expiry = DEFAULT_MIN_EXPIRY
static pthread_t	monitor_thread = AST_PTHREADT_NULL
	This is the thread for the monitor which checks for input on the channels which are not currently in use.
static const char	no_debug_usage []
static const char	no_history_usage []
static const char	notify_config [] = "sip_notify.conf"
static struct ast_config *	notify_types
static const char	notify_usage []
static int	ourport
static struct sockaddr_in	outboundproxyip
static int	pedanticsipchecking
static struct ast_peer_list	peerl
	The peer list: Peers and Friends.
static const char	prune_realtime_usage []
static int	recordhistory
static const struct c_referstatusstring	referstatusstrings []
static struct ast_register_list	regl
	The register list: Other SIP proxies we register with and place calls to.
static int	regobjs = 0
static int	rpeerobjs = 0
static int	ruserobjs = 0
static struct sched_context *	sched
static const char	show_channel_usage []
static const char	show_channels_usage []
static const char	show_domains_usage []
static const char	show_history_usage []
static const char	show_inuse_usage []
static const char	show_objects_usage []
static const char	show_peer_usage []
static const char	show_peers_usage []
static const char	show_reg_usage []
static const char	show_settings_usage []
static const char	show_subscriptions_usage []
static const char	show_user_usage []
static const char	show_users_usage []
static struct ast_custom_function	sip_header_function
static const struct cfsip_methods	sip_methods []
static const struct cfsip_options	sip_options []
	List of well-known SIP options. If we get this in a require, we should check the list and answer accordingly.
static const char	sip_reload_usage []
static int	sip_reloading = FALSE
static enum channelreloadreason	sip_reloadreason
static struct ast_rtp_protocol	sip_rtp
	Interface structure with callbacks used to connect to RTP module.
static const struct ast_channel_tech	sip_tech
	Definition of this channel for PBX channel registration.
static const struct ast_channel_tech	sip_tech_info
	This version of the sip channel tech has no send_digit_begin callback. This is for use with channels using SIP INFO DTMF so that the core knows that the channel doesn't want DTMF BEGIN frames.
static struct ast_udptl_protocol	sip_udptl
	Interface structure with callbacks used to connect to UDPTL module.
static struct ast_custom_function	sipchaninfo_function
	Structure to declare a dialplan function: SIPCHANINFO.
ast_custom_function	sippeer_function
	Structure to declare a dialplan function: SIPPEER.
static int	sipsock = -1
static int *	sipsock_read_id
static int	speerobjs = 0
static const struct cfsubscription_types	subscription_types []
static int	suserobjs = 0
static char *	synopsis_dtmfmode = "Change the dtmfmode for a SIP call"
static char *	synopsis_sipaddheader = "Add a SIP header to the outbound call"
static struct ast_user_list	userl
	The user list: Users and friends.

---

Define Documentation

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

SIP Methods we support. Definition at line 467 of file chan_sip.c. Referenced by respprep(), transmit_invite(), transmit_notify_with_sipfrag(), transmit_refer(), and transmit_reinvite_with_sdp().

#define append_history (  p, event, fmt, args...   )     append_history_full(p, "%-15s " fmt, event, ## args)

Append to SIP dialog history. Returns: Always returns 0 Definition at line 1870 of file chan_sip.c. Referenced by __sip_autodestruct(), append_send_history(), auto_congest(), build_reply_digest(), cb_extensionstate(), dialogstatechange(), do_register_auth(), handle_invite_replaces(), handle_request(), handle_request_bye(), handle_request_invite(), handle_request_refer(), handle_request_register(), handle_request_subscribe(), local_attended_transfer(), process_sip_queue(), retrans_pkt(), send_request(), send_response(), sip_cancel_destroy(), sip_fixup(), sip_hangup(), sip_new(), sip_park_thread(), sip_reregister(), sip_scheddestroy(), sip_set_rtp_peer(), transmit_register(), transmit_reinvite_with_sdp(), and transmit_response_with_auth().

#define CALLERID_UNKNOWN   "Unknown"

Definition at line 189 of file chan_sip.c.

#define DEC_CALL_LIMIT   0

Definition at line 597 of file chan_sip.c. Referenced by __sip_destroy(), handle_request_invite(), handle_response_invite(), sip_hangup(), and update_call_counter().

#define DEC_CALL_RINGING   2

Definition at line 599 of file chan_sip.c. Referenced by __sip_destroy(), handle_response_answer(), handle_response_invite(), and update_call_counter().

#define DEFAULT_ALLOW_EXT_DOM   TRUE

Definition at line 499 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_ALLOWGUEST   TRUE

Definition at line 493 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_AUTOCREATEPEER   FALSE

Definition at line 503 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_CALLERID   "asterisk"

Definition at line 490 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_COMPACTHEADERS   FALSE

Definition at line 495 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_CONTEXT   "default"

Definition at line 486 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_DEFAULT_EXPIRY   120

Definition at line 161 of file chan_sip.c. Referenced by reload_config().

#define DEFAULT_EXPIRY   900

Expire slowly Definition at line 178 of file chan_sip.c.

#define DEFAULT_FREQ_NOTOK   10 * 1000

Qualification: How often to check, if the host is down... Definition at line 193 of file chan_sip.c.

#define DEFAULT_FREQ_OK   60 * 1000

Qualification: How often to check for the host to be up Definition at line 192 of file chan_sip.c.

#define DEFAULT_MAX_CALL_BITRATE   (384)

Max bitrate for video Definition at line 506 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_MAX_EXPIRY   3600

Definition at line 163 of file chan_sip.c. Referenced by reload_config().

#define DEFAULT_MAX_FORWARDS   "70"

Definition at line 165 of file chan_sip.c. Referenced by reload_config(), reqprep(), reset_global_settings(), transmit_refer(), and transmit_register().

#define DEFAULT_MAXMS   2000

Qualification: Must be faster than 2 seconds by default Definition at line 191 of file chan_sip.c.

#define DEFAULT_MIN_EXPIRY   60

Definition at line 162 of file chan_sip.c. Referenced by reload_config().

#define DEFAULT_MOHINTERPRET   "default"

Definition at line 487 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_MOHSUGGEST   ""

Definition at line 488 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_MWITIME   10

Definition at line 492 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

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

Definition at line 491 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_NOTIFYRINGING   TRUE

Definition at line 501 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_PEDANTIC   FALSE

Definition at line 502 of file chan_sip.c. Referenced by reload_config().

#define DEFAULT_QUALIFY   FALSE

Definition at line 504 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_REALM   "asterisk"

Definition at line 500 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_REGISTRATION_TIMEOUT   20

Definition at line 164 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_RETRANS   1000

How frequently to retransmit Default: 2 * 500 ms in RFC 3261 Definition at line 195 of file chan_sip.c.

#define DEFAULT_SRVLOOKUP   FALSE

Recommended setting is ON Definition at line 494 of file chan_sip.c. Referenced by reload_config().

#define DEFAULT_T1MIN   100

100 MS for minimal roundtrip time Definition at line 505 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#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 497 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#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 496 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#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 498 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_TRANS_TIMEOUT   -1

Definition at line 201 of file chan_sip.c. Referenced by __sip_autodestruct(), cb_extensionstate(), check_auth(), check_pendings(), handle_invite_replaces(), handle_request(), handle_request_bye(), handle_request_cancel(), handle_request_info(), handle_request_invite(), handle_request_notify(), handle_request_options(), handle_request_register(), handle_request_subscribe(), handle_response_invite(), handle_response_register(), receive_message(), sip_hangup(), and sip_send_mwi_to_peer().

#define DEFAULT_USERAGENT   "Asterisk PBX"

Default Useragent: header unless re-defined in sip.conf Definition at line 508 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define DEFAULT_VMEXTEN   "asterisk"

Definition at line 489 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define EXPIRY_GUARD_LIMIT   30

Below here, we use EXPIRY_GUARD_PCT instead of EXPIRY_GUARD_SECS Definition at line 170 of file chan_sip.c.

#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 172 of file chan_sip.c.

#define EXPIRY_GUARD_PCT   0.20

Percentage of expires timeout to use when below EXPIRY_GUARD_LIMIT Definition at line 176 of file chan_sip.c.

#define EXPIRY_GUARD_SECS   15

How long before expiry do we reregister Definition at line 169 of file chan_sip.c.

#define FALSE   0

Definition at line 147 of file chan_sip.c.

#define FLAG_FATAL   (1 << 1)

Definition at line 1025 of file chan_sip.c. Referenced by __sip_reliable_xmit(), and retrans_pkt().

#define FLAG_RESPONSE   (1 << 0)

Definition at line 1024 of file chan_sip.c. Referenced by __sip_ack(), __sip_pretend_ack(), __sip_reliable_xmit(), __sip_semi_ack(), handle_request(), and retrans_pkt().

#define FORMAT   "%-15.15s %-10.10s %-11.11s %5.5d/%5.5d %-4.4s %-3.3s %-3.3s %-15.15s %-10.10s\n"

#define FORMAT   "%-30.30s %-12.12s %8d %-20.20s %-25.25s\n"

#define FORMAT   "%-40.40s %-20.20s %-16.16s\n"

#define FORMAT   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %-10s\n"

#define FORMAT   "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"

#define FORMAT   "%-25.25s %-15.15s %-15.15s \n"

#define FORMAT2   "%-15.15s %-10.10s %-11.11s %-11.11s %-4.4s %-7.7s %-15.15s\n"

#define FORMAT2   "%-30.30s %-12.12s %8.8s %-20.20s %-25.25s\n"

#define FORMAT2   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %-10s\n"

#define FORMAT2   "%-25.25s %-15.15s %-15.15s \n"

#define FORMAT3   "%-15.15s %-10.10s %-11.11s %-15.15s %-13.13s %-15.15s %-10.10s\n"

Referenced by __sip_show_channels().

#define INC_CALL_LIMIT   1

Definition at line 598 of file chan_sip.c. Referenced by handle_request_invite(), sip_hangup(), and update_call_counter().

#define INC_CALL_RINGING   3

Definition at line 600 of file chan_sip.c. Referenced by sip_call(), and update_call_counter().

#define INITIAL_CSEQ   101

our initial sip sequence number Definition at line 208 of file chan_sip.c. Referenced by sip_alloc(), sip_register(), and transmit_response_using_temp().

#define IPTOS_MINCOST   0x02

Definition at line 156 of file chan_sip.c.

#define MAX (  a, b   )     ((a) > (b) ? (a) : (b))

Definition at line 186 of file chan_sip.c.

#define MAX_AUTHTRIES   3

Try authentication three times, then fail Definition at line 202 of file chan_sip.c. Referenced by handle_response(), handle_response_bye(), handle_response_invite(), and handle_response_register().

#define MAX_RETRANS   6

Try only 6 times for retransmissions, a total of 7 transmissions Definition at line 196 of file chan_sip.c.

#define NO_RTP   0

Definition at line 224 of file chan_sip.c.

#define NOT_SUPPORTED   0

Definition at line 400 of file chan_sip.c.

#define RTP   1

Definition at line 223 of file chan_sip.c.

#define SDP_SAMPLE_RATE (  x   )     (x == AST_FORMAT_G722) ? 16000 : 8000

Definition at line 6230 of file chan_sip.c. Referenced by add_sdp().

#define SIP_ALREADYGONE   (1 << 0)

Whether or not we've already been destroyed by our peer Definition at line 703 of file chan_sip.c. Referenced by handle_request(), handle_request_bye(), handle_request_cancel(), handle_request_invite(), handle_request_refer(), handle_response(), handle_response_invite(), retrans_pkt(), sip_alreadygone(), sip_hangup(), and sip_set_rtp_peer().

#define SIP_CALL_LIMIT   (1 << 28)

Call limit enforced for this call Definition at line 744 of file chan_sip.c. Referenced by check_peer_ok(), check_user_full(), check_user_ok(), create_addr_from_peer(), and update_call_counter().

#define SIP_CAN_REINVITE   (1 << 20)

allow peers to be reinvited to send media directly p2p Definition at line 732 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), handle_common_options(), reload_config(), reset_global_settings(), sip_get_rtp_peer(), sip_get_udptl_peer(), sip_get_vrtp_peer(), and sip_handle_t38_reinvite().

#define SIP_CAN_REINVITE_NAT   (2 << 20)

allow media reinvite when new peer is behind NAT Definition at line 733 of file chan_sip.c. Referenced by handle_common_options(), sip_get_rtp_peer(), and sip_set_rtp_peer().

#define SIP_DEFER_BYE_ON_TRANSFER   (1 << 15)

Do not hangup at first ast_hangup Definition at line 718 of file chan_sip.c. Referenced by handle_invite_replaces(), handle_request_refer(), local_attended_transfer(), and sip_hangup().

#define SIP_DTMF   (3 << 16)

DTMF Support: four settings, uses two bits Definition at line 719 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), check_peer_ok(), check_user_full(), check_user_ok(), create_addr_from_peer(), dialog_activate_media(), handle_common_options(), handle_request_invite(), sip_alloc(), sip_dtmfmode(), sip_new(), sip_rtp_read(), sip_senddigit_begin(), sip_senddigit_end(), sip_show_channel(), and sip_show_settings().

#define SIP_DTMF_AUTO   (3 << 16)

DTMF Support: AUTO switch between rfc2833 and in-band DTMF Definition at line 723 of file chan_sip.c. Referenced by check_peer_ok(), check_user_full(), check_user_ok(), create_addr_from_peer(), dtmfmode2str(), handle_common_options(), and sip_alloc().

#define SIP_DTMF_INBAND   (1 << 16)

DTMF Support: Inband audio, only for ULAW/ALAW - "inband" Definition at line 721 of file chan_sip.c. Referenced by dtmfmode2str(), handle_common_options(), sip_dtmfmode(), sip_new(), sip_rtp_read(), sip_senddigit_begin(), and sip_senddigit_end().

#define SIP_DTMF_INFO   (2 << 16)

DTMF Support: SIP Info messages - "info" Definition at line 722 of file chan_sip.c. Referenced by create_addr_from_peer(), dialog_activate_media(), dtmfmode2str(), handle_common_options(), handle_request_invite(), sip_dtmfmode(), sip_new(), and sip_senddigit_end().

#define SIP_DTMF_RFC2833   (0 << 16)

DTMF Support: RTP DTMF - "rfc2833" Definition at line 720 of file chan_sip.c. Referenced by check_peer_ok(), check_user_full(), check_user_ok(), create_addr_from_peer(), dtmfmode2str(), handle_common_options(), handle_request_invite(), reload_config(), reset_global_settings(), sip_alloc(), sip_dtmfmode(), sip_rtp_read(), sip_senddigit_begin(), and sip_senddigit_end().

#define SIP_FLAGS_TO_COPY

Value: (SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \ SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \ SIP_USEREQPHONE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE) Definition at line 749 of file chan_sip.c. Referenced by build_user(), check_peer_ok(), check_user_full(), check_user_ok(), create_addr_from_peer(), set_device_defaults(), set_peer_defaults(), sip_alloc(), and sip_poke_peer().

#define SIP_FREE_BIT   (1 << 14)

---- Definition at line 717 of file chan_sip.c.

#define SIP_G726_NONSTANDARD   (1 << 31)

Use non-standard packing for G726-32 data Definition at line 747 of file chan_sip.c. Referenced by add_codec_to_sdp(), and handle_common_options().

#define SIP_GOTREFER   (1 << 7)

Got a refer? Definition at line 710 of file chan_sip.c. Referenced by handle_request_refer(), local_attended_transfer(), sip_handle_t38_reinvite(), sip_read(), sip_set_rtp_peer(), and sip_set_udptl_peer().

#define SIP_INC_COUNT   (1 << 30)

Did this connection increment the counter of in-use calls? Definition at line 746 of file chan_sip.c. Referenced by __sip_destroy(), and update_call_counter().

#define SIP_INSECURE_INVITE   (1 << 24)

don't require authentication for incoming INVITEs Definition at line 737 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), check_peer_ok(), check_user_full(), and handle_common_options().

#define SIP_INSECURE_PORT   (1 << 23)

don't require matching port for incoming requests Definition at line 736 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), handle_common_options(), and sip_addrcmp().

#define SIP_MAX_HEADERS   64

Max amount of SIP headers to read Definition at line 204 of file chan_sip.c. Referenced by add_header(), and parse_request().

#define SIP_MAX_LINES   64

Max amount of lines in SIP attachment (like SDP) Definition at line 205 of file chan_sip.c. Referenced by add_line(), and parse_request().

#define SIP_MAX_PACKET   4096

Also from RFC 3261 (2543), should sub headers tho Definition at line 206 of file chan_sip.c. Referenced by __transmit_response(), transmit_info_with_digit(), transmit_info_with_vidupdate(), transmit_invite(), transmit_message_with_text(), transmit_notify_with_mwi(), transmit_notify_with_sipfrag(), transmit_refer(), transmit_register(), transmit_reinvite_with_sdp(), transmit_request(), transmit_request_with_auth(), transmit_response_with_attachment(), transmit_response_with_auth(), transmit_response_with_unsupported(), and transmit_state_notify().

#define SIP_NAT   (3 << 18)

four settings, uses two bits Definition at line 725 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), build_via(), copy_via_headers(), create_addr_from_peer(), handle_common_options(), register_verify(), sip_alloc(), sip_nat_mode(), sip_real_dst(), sip_show_channel(), sip_show_settings(), and transmit_response_using_temp().

#define SIP_NAT_ALWAYS   (3 << 18)

NAT Both ROUTE and RFC3581 Definition at line 729 of file chan_sip.c. Referenced by copy_via_headers(), handle_common_options(), and nat2str().

#define SIP_NAT_NEVER   (0 << 18)

No nat support Definition at line 726 of file chan_sip.c. Referenced by handle_common_options(), and nat2str().

#define SIP_NAT_RFC3581   (1 << 18)

NAT RFC3581 Definition at line 727 of file chan_sip.c. Referenced by build_via(), copy_via_headers(), handle_common_options(), nat2str(), reload_config(), and reset_global_settings().

#define SIP_NAT_ROUTE   (2 << 18)

NAT Only ROUTE Definition at line 728 of file chan_sip.c. Referenced by _sip_show_devices(), _sip_show_peers(), check_peer_ok(), check_user_full(), check_user_ok(), check_via(), create_addr_from_peer(), handle_common_options(), nat2str(), parse_register_contact(), send_request(), set_address_from_contact(), sip_alloc(), sip_nat_mode(), sip_real_dst(), and transmit_response_using_temp().

#define SIP_NEEDDESTROY   (1 << 1)

if we need to be destroyed by the monitor thread Definition at line 704 of file chan_sip.c. Referenced by __sip_show_channels(), do_monitor(), handle_request(), handle_request_refer(), handle_request_subscribe(), handle_response(), handle_response_bye(), handle_response_invite(), handle_response_message(), handle_response_notify(), handle_response_peerpoke(), handle_response_refer(), handle_response_register(), retrans_pkt(), sip_hangup(), sip_reg_timeout(), and sip_show_channel().

#define SIP_NEEDREINVITE   (1 << 5)

Do we need to send another reinvite? Definition at line 708 of file chan_sip.c. Referenced by check_pendings(), sip_handle_t38_reinvite(), sip_hangup(), sip_read(), sip_set_rtp_peer(), and sip_set_udptl_peer().

#define SIP_NO_HISTORY   (1 << 27)

Suppress recording request/response history Definition at line 743 of file chan_sip.c. Referenced by do_register_auth(), handle_request_bye(), handle_request_invite(), process_sip_queue(), send_request(), send_response(), sip_alloc(), sip_hangup(), sip_new(), sip_reregister(), sip_scheddestroy(), sip_set_rtp_peer(), temp_pvt_init(), transmit_register(), transmit_reinvite_with_sdp(), and transmit_response_using_temp().

#define SIP_NOVIDEO   (1 << 2)

Didn't get video in invite, don't offer Definition at line 705 of file chan_sip.c. Referenced by add_sdp(), process_sdp(), and sip_indicate().

#define SIP_OPT_100REL   (1 << 1)

Definition at line 403 of file chan_sip.c.

#define SIP_OPT_EARLY_SESSION   (1 << 3)

Definition at line 405 of file chan_sip.c.

#define SIP_OPT_EVENTLIST   (1 << 11)

Definition at line 413 of file chan_sip.c.

#define SIP_OPT_GRUU   (1 << 12)

Definition at line 414 of file chan_sip.c.

#define SIP_OPT_HISTINFO   (1 << 15)

Definition at line 417 of file chan_sip.c.

#define SIP_OPT_JOIN   (1 << 4)

Definition at line 406 of file chan_sip.c.

#define SIP_OPT_NOREFERSUB   (1 << 14)

Definition at line 416 of file chan_sip.c.

#define SIP_OPT_PATH   (1 << 5)

Definition at line 407 of file chan_sip.c.

#define SIP_OPT_PRECONDITION   (1 << 7)

Definition at line 409 of file chan_sip.c.

#define SIP_OPT_PREF   (1 << 6)

Definition at line 408 of file chan_sip.c.

#define SIP_OPT_PRIVACY   (1 << 8)

Definition at line 410 of file chan_sip.c.

#define SIP_OPT_REPLACES   (1 << 0)

Definition at line 402 of file chan_sip.c. Referenced by handle_request_invite().

#define SIP_OPT_RESPRIORITY   (1 << 16)

Definition at line 418 of file chan_sip.c.

#define SIP_OPT_SDP_ANAT   (1 << 9)

Definition at line 411 of file chan_sip.c.

#define SIP_OPT_SEC_AGREE   (1 << 10)

Definition at line 412 of file chan_sip.c.

#define SIP_OPT_TARGET_DIALOG   (1 << 13)

Definition at line 415 of file chan_sip.c.

#define SIP_OPT_TIMER   (1 << 2)

Definition at line 404 of file chan_sip.c.

#define SIP_OUTGOING   (1 << 13)

Direction of the last transaction in this dialog Definition at line 716 of file chan_sip.c. Referenced by handle_request_bye(), handle_request_invite(), handle_request_refer(), handle_request_subscribe(), handle_response(), handle_response_invite(), reqprep(), respprep(), sip_call(), sip_hangup(), sip_indicate(), sip_poke_peer(), sip_send_mwi_to_peer(), sip_show_channel(), sip_write(), transmit_refer(), transmit_register(), transmit_reinvite_with_sdp(), transmit_response_with_attachment(), transmit_response_with_sdp(), update_call_counter(), and write_media_frame().

#define SIP_PAGE2_ALLOWOVERLAP   (1 << 17)

Allow overlap dialing ? Definition at line 770 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), get_destination(), handle_common_options(), handle_request_invite(), reload_config(), reset_global_settings(), and sip_show_settings().

#define SIP_PAGE2_ALLOWSUBSCRIBE   (1 << 16)

Allow subscriptions from this peer? Definition at line 769 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), build_device(), build_peer(), handle_common_options(), handle_request_subscribe(), reload_config(), reset_global_settings(), and sip_show_settings().

#define SIP_PAGE2_BUGGY_MWI   (1 << 26)

26: Buggy CISCO MWI fix Definition at line 781 of file chan_sip.c. Referenced by handle_common_options(), and transmit_notify_with_mwi().

#define SIP_PAGE2_CALL_ONHOLD   (3 << 23)

Call states Definition at line 777 of file chan_sip.c. Referenced by __sip_show_channels(), and update_call_counter().

#define SIP_PAGE2_CALL_ONHOLD_INACTIVE   (1 << 24)

24: Inactive Definition at line 779 of file chan_sip.c. Referenced by add_sdp().

#define SIP_PAGE2_CALL_ONHOLD_ONEDIR   (1 << 23)

23: One directional hold Definition at line 778 of file chan_sip.c. Referenced by add_sdp().

#define SIP_PAGE2_DEBUG   (3 << 11)

Definition at line 763 of file chan_sip.c.

#define SIP_PAGE2_DEBUG_CONFIG   (1 << 11)

Definition at line 764 of file chan_sip.c. Referenced by reload_config(), and reset_global_settings().

#define SIP_PAGE2_DEBUG_CONSOLE   (1 << 12)

Definition at line 765 of file chan_sip.c. Referenced by reload_config(), sip_do_debug(), sip_do_debug_device(), sip_do_debug_ip(), sip_do_debug_peer(), and sip_no_debug().

#define SIP_PAGE2_DYNAMIC   (1 << 13)

Dynamic Peers register with Asterisk Definition at line 766 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_devices(), _sip_show_peer(), _sip_show_peers(), build_device(), build_peer(), function_sippeer(), register_verify(), set_device_host(), temp_device(), and temp_peer().

#define SIP_PAGE2_FLAGS_TO_COPY

Value: (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \ SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI) Definition at line 783 of file chan_sip.c. Referenced by build_user(), check_peer_ok(), check_user_full(), check_user_ok(), create_addr_from_peer(), set_device_defaults(), set_peer_defaults(), sip_alloc(), and sip_poke_peer().

#define SIP_PAGE2_IGNOREREGEXPIRE   (1 << 10)

Definition at line 762 of file chan_sip.c. Referenced by build_device(), build_peer(), destroy_association(), reload_config(), and sip_show_settings().

#define SIP_PAGE2_INC_RINGING   (1 << 19)

Did this connection increment the counter of in-use calls? Definition at line 772 of file chan_sip.c. Referenced by __sip_destroy(), and update_call_counter().

#define SIP_PAGE2_RFC2833_COMPENSATE   (1 << 25)

25: ???? Definition at line 780 of file chan_sip.c. Referenced by create_addr_from_peer(), dialog_activate_media(), handle_common_options(), handle_request_invite(), sip_alloc(), and sip_show_settings().

#define SIP_PAGE2_RT_FROMCONTACT   (1 << 4)

Definition at line 759 of file chan_sip.c. Referenced by build_device(), build_peer(), destroy_association(), parse_register_contact(), and reg_source_db().

#define SIP_PAGE2_RTAUTOCLEAR   (1 << 2)

Definition at line 758 of file chan_sip.c. Referenced by expire_register(), and reload_config().

#define SIP_PAGE2_RTCACHEFRIENDS   (1 << 0)

Definition at line 756 of file chan_sip.c. Referenced by complete_sip_prune_realtime_peer(), complete_sip_prune_realtime_user(), reload_config(), sip_prune_realtime(), sip_show_settings(), and update_peer().

#define SIP_PAGE2_RTSAVE_SYSNAME   (1 << 5)

Definition at line 760 of file chan_sip.c. Referenced by realtime_update_peer(), reload_config(), and sip_show_settings().

#define SIP_PAGE2_RTUPDATE   (1 << 1)

Definition at line 757 of file chan_sip.c. Referenced by reload_config(), reset_global_settings(), sip_show_settings(), and update_peer().

#define SIP_PAGE2_SELFDESTRUCT   (1 << 14)

Automatic peers need to destruct themselves Definition at line 767 of file chan_sip.c. Referenced by expire_register(), sip_destroy_device(), sip_destroy_peer(), temp_device(), and temp_peer().

#define SIP_PAGE2_SUBSCRIBEMWIONLY   (1 << 18)

Only issue MWI notification if subscribed to Definition at line 771 of file chan_sip.c. Referenced by build_device(), build_peer(), does_peer_need_mwi(), and register_verify().

#define SIP_PAGE2_T38SUPPORT   (7 << 20)

T38 Fax Passthrough Support Definition at line 773 of file chan_sip.c. Referenced by create_addr_from_peer(), dialog_activate_media(), and sip_alloc().

#define SIP_PAGE2_T38SUPPORT_RTP   (2 << 20)

21: T38 Fax Passthrough Support (not implemented) Definition at line 775 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), add_sdp(), handle_common_options(), and sip_show_settings().

#define SIP_PAGE2_T38SUPPORT_TCP   (4 << 20)

22: T38 Fax Passthrough Support (not implemented) Definition at line 776 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), handle_common_options(), and sip_show_settings().

#define SIP_PAGE2_T38SUPPORT_UDPTL   (1 << 20)

20: T38 Fax Passthrough Support Definition at line 774 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), handle_common_options(), sip_read(), sip_rtp_read(), and sip_show_settings().

#define SIP_PAGE2_VIDEOSUPPORT   (1 << 15)

Definition at line 768 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_devices(), _sip_show_peer(), _sip_show_peers(), check_peer_ok(), check_user_full(), check_user_ok(), create_addr_from_peer(), dialog_activate_media(), handle_common_options(), reload_config(), sip_alloc(), and sip_show_settings().

#define SIP_PENDINGBYE   (1 << 6)

Need to send bye after we ack? Definition at line 709 of file chan_sip.c. Referenced by check_pendings(), handle_response_answer(), handle_response_invite(), sip_handle_t38_reinvite(), sip_hangup(), sip_read(), sip_set_rtp_peer(), and sip_set_udptl_peer().

#define SIP_PKT_DEBUG   (1 << 0)

Debug this packet Definition at line 788 of file chan_sip.c. Referenced by find_via_branch(), handle_request(), handle_request_message(), handle_request_refer(), handle_request_register(), handle_request_subscribe(), handle_response(), initialize_initreq(), and sipsock_read().

#define SIP_PKT_IGNORE   (1 << 2)

This is a re-transmit, ignore it Definition at line 790 of file chan_sip.c. Referenced by check_peer_ok(), check_user_full(), check_user_ok(), handle_invite_replaces(), handle_request(), handle_request_bye(), handle_request_info(), handle_request_invite(), handle_request_message(), handle_request_refer(), handle_request_subscribe(), handle_response(), handle_response_answer(), handle_response_invite(), and register_verify().

#define SIP_PKT_WITH_TOTAG   (1 << 1)

This packet has a to-tag Definition at line 789 of file chan_sip.c. Referenced by find_call(), handle_request(), and match_or_create_dialog().

#define SIP_PROG_INBAND   (3 << 25)

three settings, uses two bits Definition at line 739 of file chan_sip.c. Referenced by handle_common_options(), sip_indicate(), and sip_show_settings().

#define SIP_PROG_INBAND_NEVER   (0 << 25)

Definition at line 740 of file chan_sip.c. Referenced by sip_indicate(), and sip_show_settings().

#define SIP_PROG_INBAND_NO   (1 << 25)

Definition at line 741 of file chan_sip.c. Referenced by handle_common_options(), and sip_show_settings().

#define SIP_PROG_INBAND_YES   (2 << 25)

Definition at line 742 of file chan_sip.c. Referenced by handle_common_options(), and sip_indicate().

#define SIP_PROGRESS_SENT   (1 << 4)

Have sent 183 message progress Definition at line 707 of file chan_sip.c. Referenced by sip_indicate(), sip_write(), and write_media_frame().

#define SIP_PROMISCREDIR   (1 << 8)

Promiscuous redirection Definition at line 711 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), handle_common_options(), parse_moved_contact(), sip_show_channel(), and sip_show_settings().

#define SIP_REALTIME   (1 << 11)

Flag for realtime users Definition at line 714 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_devices(), _sip_show_peer(), _sip_show_peers(), build_device(), build_peer(), expire_register(), parse_register_contact(), sip_destroy_device(), sip_destroy_peer(), sip_destroy_user(), and update_peer().

#define SIP_REINVITE   (7 << 20)

three bits used Definition at line 731 of file chan_sip.c. Referenced by handle_common_options().

#define SIP_REINVITE_UPDATE   (4 << 20)

use UPDATE (RFC3311) when reinviting this peer Definition at line 734 of file chan_sip.c. Referenced by handle_common_options(), and transmit_reinvite_with_sdp().

#define SIP_RINGING   (1 << 3)

Have sent 180 ringing Definition at line 706 of file chan_sip.c. Referenced by sip_indicate().

#define SIP_SENDRPID   (1 << 29)

Remote Party-ID Support Definition at line 745 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), and handle_common_options().

#define SIP_TIMER_T1   500

Definition at line 197 of file chan_sip.c. Referenced by create_addr(), sip_alloc(), and sip_scheddestroy().

#define SIP_TRANS_TIMEOUT   32000

SIP request timeout (rfc 3261) 64*T1 Todo: Use known T1 for timeout (peerpoke) Definition at line 198 of file chan_sip.c. Referenced by sip_call().

#define SIP_TRUSTRPID   (1 << 9)

Trust RPID headers? Definition at line 712 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), handle_common_options(), and replace_cid().

#define SIP_USECLIENTCODE   (1 << 12)

Trust X-ClientCode info message Definition at line 715 of file chan_sip.c. Referenced by handle_common_options(), handle_request_info(), and sip_show_settings().

#define SIP_USEREQPHONE   (1 << 10)

Add user=phone to numeric URI. Default off Definition at line 713 of file chan_sip.c. Referenced by _sip_show_device(), _sip_show_peer(), build_device(), build_peer(), initreqprep(), reload_config(), and sip_show_settings().

#define sipdebug   ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)

Definition at line 818 of file chan_sip.c. Referenced by __sip_ack(), __sip_reliable_xmit(), __sip_semi_ack(), device_ref(), device_unref(), dialog_lock(), dialoglist_lock(), dialoglist_unlock(), dialogstatechange(), handle_request_info(), handle_request_invite(), handle_request_notify(), handle_request_refer(), handle_request_subscribe(), local_attended_transfer(), make_our_tag(), parse_request(), parse_sip_options(), reqprep(), retrans_pkt(), sip_addheader(), sip_call(), sip_debug_test_addr(), sip_debug_test_pvt(), sip_dump_history(), sip_hangup(), sip_option_lookup(), sip_poke_peer(), sip_register(), sip_reregister(), sipnet_lock(), sipnet_unlock(), stop_media_flows(), transmit_invite(), transmit_refer(), transmit_register(), transmit_reinvite_with_sdp(), and update_call_counter().

#define sipdebug_config   ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)

Definition at line 819 of file chan_sip.c.

#define sipdebug_console   ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)

Definition at line 820 of file chan_sip.c. Referenced by sip_do_debug().

#define STANDARD_SIP_PORT   5060

Standard SIP port from RFC 3261. DO NOT CHANGE THIS. Definition at line 473 of file chan_sip.c. Referenced by build_contact(), build_peer(), check_via(), create_addr(), parse_register_contact(), reload_config(), reset_ip_interface(), set_address_from_contact(), set_destination(), set_device_defaults(), set_device_host(), set_peer_defaults(), and sip_show_registry().

#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 399 of file chan_sip.c.

#define SUPPORTED_EXTENSIONS   "replaces"

SIP Extensions we support. Definition at line 470 of file chan_sip.c. Referenced by respprep(), transmit_invite(), transmit_notify_with_sipfrag(), transmit_refer(), and transmit_reinvite_with_sdp().

#define T38FAX_FILL_BIT_REMOVAL   (1 << 0)

Default: 0 (unset) Definition at line 793 of file chan_sip.c. Referenced by add_t38_sdp().

#define T38FAX_RATE_12000   (1 << 12)

12000 bps t38FaxRate Definition at line 812 of file chan_sip.c. Referenced by t38_get_rate().

#define T38FAX_RATE_14400   (1 << 13)

14400 bps t38FaxRate This is default: NO MMR and JBIG transcoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate Definition at line 813 of file chan_sip.c. Referenced by t38_get_rate().

#define T38FAX_RATE_2400   (1 << 8)

2400 bps t38FaxRate Definition at line 808 of file chan_sip.c. Referenced by reset_global_settings(), and t38_get_rate().

#define T38FAX_RATE_4800   (1 << 9)

4800 bps t38FaxRate Definition at line 809 of file chan_sip.c. Referenced by reset_global_settings(), and t38_get_rate().

#define T38FAX_RATE_7200   (1 << 10)

7200 bps t38FaxRate Definition at line 810 of file chan_sip.c. Referenced by reset_global_settings(), and t38_get_rate().

#define T38FAX_RATE_9600   (1 << 11)

9600 bps t38FaxRate Definition at line 811 of file chan_sip.c. Referenced by reset_global_settings(), and t38_get_rate().

#define T38FAX_RATE_MANAGEMENT_LOCAL_TCF   (1 << 3)

Unset for transferredTCF (UDPTL), set for localTCF (TPKT) Definition at line 798 of file chan_sip.c. Referenced by add_t38_sdp().

#define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF   (0 << 3)

Definition at line 797 of file chan_sip.c. Referenced by create_addr_from_peer(), dialog_activate_media(), and sip_alloc().

#define T38FAX_TRANSCODING_JBIG   (1 << 2)

Default: 0 (unset) Definition at line 795 of file chan_sip.c. Referenced by add_t38_sdp().

#define T38FAX_TRANSCODING_MMR   (1 << 1)

Default: 0 (unset) Definition at line 794 of file chan_sip.c. Referenced by add_t38_sdp().

#define T38FAX_UDP_EC_FEC   (1 << 4)

Set for t38UDPFEC Definition at line 801 of file chan_sip.c. Referenced by create_addr_from_peer(), dialog_activate_media(), and sip_alloc().

#define T38FAX_UDP_EC_NONE   (0 << 4)

two bits, if unset NO t38UDPEC field in T38 SDP Definition at line 800 of file chan_sip.c. Referenced by add_t38_sdp(), create_addr_from_peer(), dialog_activate_media(), and sip_alloc().

#define T38FAX_UDP_EC_REDUNDANCY   (2 << 4)

Set for t38UDPRedundancy Definition at line 802 of file chan_sip.c. Referenced by add_t38_sdp(), create_addr_from_peer(), dialog_activate_media(), and sip_alloc().

#define T38FAX_VERSION   (3 << 6)

two bits, 2 values so far, up to 4 values max Definition at line 804 of file chan_sip.c. Referenced by add_t38_sdp().

#define T38FAX_VERSION_0   (0 << 6)

Version 0 Definition at line 805 of file chan_sip.c. Referenced by add_t38_sdp(), and reset_global_settings().

#define T38FAX_VERSION_1   (1 << 6)

Version 1 Definition at line 806 of file chan_sip.c. Referenced by add_t38_sdp().

#define TRUE   1

Definition at line 151 of file chan_sip.c.

#define UNLINK (  element, head, prev   )

--- some list management macros. Definition at line 1594 of file chan_sip.c. Referenced by __sip_ack(), __sip_destroy(), and retrans_pkt().

#define VIDEO_CODEC_MASK   0x1fc0000

Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO Definition at line 154 of file chan_sip.c.

---

Enumeration Type Documentation

enum can_create_dialog

Enumerator: CAN_NOT_CREATE_DIALOG  CAN_CREATE_DIALOG  CAN_CREATE_DIALOG_UNSUPPORTED_METHOD  Definition at line 357 of file chan_sip.c. { CAN_NOT_CREATE_DIALOG, CAN_CREATE_DIALOG, CAN_CREATE_DIALOG_UNSUPPORTED_METHOD, };

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  AUTH_FAKE_AUTH  AUTH_UNKNOWN_DOMAIN  Definition at line 332 of file chan_sip.c. { 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, AUTH_SECRET_FAILED = -1, AUTH_USERNAME_MISMATCH = -2, AUTH_NOT_FOUND = -3, /* returned by register_verify */ AUTH_FAKE_AUTH = -4, AUTH_UNKNOWN_DOMAIN = -5, };

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 666 of file chan_sip.c. { SIP_DOMAIN_AUTO, /*!< This domain is auto-configured */ SIP_DOMAIN_CONFIG, /*!< This domain is from configuration */ };

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 243 of file chan_sip.c. { INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */ INV_CALLING = 1, /*!< Invite sent, no answer */ INV_PROCEEDING = 2, /*!< We got/sent 1xx message */ INV_EARLY_MEDIA = 3, /*!< We got 18x message with to-tag back */ INV_COMPLETED = 4, /*!< Got final response with error. Wait for ACK, then CONFIRMED */ INV_CONFIRMED = 5, /*!< Confirmed response - we've got an ack (Incoming calls only) */ INV_TERMINATED = 6, /*!< 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 = 7, /*!< Transaction cancelled by client or server in non-terminated state */ };

enum parse_register_result

Enumerator: PARSE_REGISTER_FAILED  PARSE_REGISTER_UPDATE  PARSE_REGISTER_QUERY  Definition at line 270 of file chan_sip.c. { PARSE_REGISTER_FAILED, PARSE_REGISTER_UPDATE, PARSE_REGISTER_QUERY, };

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 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 842 of file chan_sip.c. { 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 */ 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 */ };

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 326 of file chan_sip.c. { PROXY_AUTH = 407, WWW_AUTH = 401, };

enum sip_result

Enumerator: AST_SUCCESS  AST_FAILURE  Definition at line 235 of file chan_sip.c. { AST_SUCCESS = 0, AST_FAILURE = -1, };

enum sipmethod

SIP Request methods known by Asterisk. Enumerator: SIP_UNKNOWN  SIP_RESPONSE  SIP_REGISTER  SIP_OPTIONS  SIP_NOTIFY  SIP_INVITE  SIP_ACK  SIP_PRACK  SIP_BYE