path: root/channels/chan_sip.c

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2006, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*!
 * \file
 * \brief Implementation of Session Initiation Protocol
 *
 * \author Mark Spencer <markster@digium.com>
 *
 * See Also:
 * \arg \ref AstCREDITS
 *
 * Implementation of RFC 3261 - without S/MIME, TCP and TLS support
 * Configuration file \link Config_sip sip.conf \endlink
 *
 *
 * \todo Better support of forking
 * \todo VIA branch tag transaction checking
 * \todo Transaction support
 * \todo We need to test TCP sessions with SIP proxies and in regards
 *       to the SIP outbound specs.
 *
 * \ingroup channel_drivers
 *
 * \par 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"
 *
 * \par 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_incoming(), that parses a bit more.
 * If it is a response to an outbound request, the packet is sent to handle_response().
 * If it is a request, handle_incoming() 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 
 * 
 * \par Outbound calls
 * Outbound calls are set up by the PBX through the sip_request_call()
 * function. After that, they are activated by sip_call().
 * 
 * \par Hanging up
 * The PBX issues a hangup on both incoming and outgoing calls through
 * the sip_hangup() function
 */

/*** MODULEINFO
        <depend>chan_local</depend>
 ***/

/*!  \page sip_session_timers SIP Session Timers in Asterisk Chan_sip

	The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
	refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
	request at a negotiated interval. If a session refresh fails then all the entities that support Session-
	Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
	the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
	that do not support Session-Timers).

	The Session-Timers can be configured on a system-wide, per-user, or per-peer basis. The peruser/
	per-peer settings override the global settings. The following new parameters have been
	added to the sip.conf file.
		session-timers=["accept", "originate", "refuse"]
		session-expires=[integer]
		session-minse=[integer]
		session-refresher=["uas", "uac"]

	The session-timers parameter in sip.conf defines the mode of operation of SIP session-timers feature in
	Asterisk. The Asterisk can be configured in one of the following three modes:

	1. Accept :: In the "accept" mode, the Asterisk server honors session-timers requests
		made by remote end-points. A remote end-point can request Asterisk to engage
		session-timers by either sending it an INVITE request with a "Supported: timer"
		header in it or by responding to Asterisk's INVITE with a 200 OK that contains
		Session-Expires: header in it. In this mode, the Asterisk server does not 
		request session-timers from remote end-points. This is the default mode.
	2. Originate :: In the "originate" mode, the Asterisk server requests the remote 
		end-points to activate session-timers in addition to honoring such requests
		made by the remote end-pints. In order to get as much protection as possible
		against hanging SIP channels due to network or end-point failures, Asterisk
		resends periodic re-INVITEs even if a remote end-point does not support
		the session-timers feature.
	3. Refuse :: In the "refuse" mode, Asterisk acts as if it does not support session-
		timers for inbound or outbound requests. If a remote end-point requests
		session-timers in a dialog, then Asterisk ignores that request unless it's
		noted as a requirement (Require: header), in which case the INVITE is 
		rejected with a 420 Bad Extension response.

*/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#include <ctype.h>
#include <sys/ioctl.h>
#include <fcntl.h>
#include <signal.h>
#include <sys/signal.h>
#include <regex.h>

#include "asterisk/network.h"
#include "asterisk/paths.h"	/* need ast_config_AST_SYSTEM_NAME */

#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.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/astobj2.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
#include "asterisk/linkedlists.h"
#include "asterisk/stringfields.h"
#include "asterisk/monitor.h"
#include "asterisk/netsock.h"
#include "asterisk/localtime.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/threadstorage.h"
#include "asterisk/translate.h"
#include "asterisk/ast_version.h"
#include "asterisk/event.h"
#include "asterisk/tcptls.h"
#include "asterisk/strings.h"

#ifndef FALSE
#define FALSE    0
#endif

#ifndef TRUE
#define TRUE     1
#endif

#define	SIPBUFSIZE		512

#define XMIT_ERROR		-2

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

/* #define VOCAL_DATA_HACK */

#define DEFAULT_DEFAULT_EXPIRY  120
#define DEFAULT_MIN_EXPIRY      60
#define DEFAULT_MAX_EXPIRY      3600
#define DEFAULT_REGISTRATION_TIMEOUT 20
#define DEFAULT_MAX_FORWARDS    "70"

/* guard limit must be larger than guard secs */
/* guard min must be < 1000, and should be >= 250 */
#define EXPIRY_GUARD_SECS       15                /*!< How long before expiry do we reregister */
#define EXPIRY_GUARD_LIMIT      30                /*!< Below here, we use EXPIRY_GUARD_PCT instead of 
	                                                 EXPIRY_GUARD_SECS */
#define EXPIRY_GUARD_MIN        500                /*!< This is the minimum guard time applied. If 
                                                   GUARD_PCT turns out to be lower than this, it 
                                                   will use this time instead.
                                                   This is in milliseconds. */
#define EXPIRY_GUARD_PCT        0.20                /*!< Percentage of expires timeout to use when 
                                                    below EXPIRY_GUARD_LIMIT */
#define DEFAULT_EXPIRY 900                          /*!< Expire slowly */

static int min_expiry = DEFAULT_MIN_EXPIRY;        /*!< Minimum accepted registration time */
static int max_expiry = DEFAULT_MAX_EXPIRY;        /*!< Maximum accepted registration time */
static int default_expiry = DEFAULT_DEFAULT_EXPIRY;

#define CALLERID_UNKNOWN             "Anonymous"
#define FROMDOMAIN_INVALID           "anonymous.invalid"

#define DEFAULT_MAXMS                2000             /*!< Qualification: Must be faster than 2 seconds by default */
#define DEFAULT_QUALIFYFREQ          60 * 1000        /*!< Qualification: How often to check for the host to be up */
#define DEFAULT_FREQ_NOTOK           10 * 1000        /*!< Qualification: How often to check, if the host is down... */

#define DEFAULT_RETRANS              1000             /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
#define MAX_RETRANS                  6                /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
#define SIP_TIMER_T1                 500              /* SIP timer T1 (according to RFC 3261) */
#define SIP_TRANS_TIMEOUT            64 * SIP_TIMER_T1/*!< SIP request timeout (rfc 3261) 64*T1 
                                                      \todo Use known T1 for timeout (peerpoke)
                                                      */
#define DEFAULT_TRANS_TIMEOUT        -1               /* Use default SIP transaction timeout */
#define PROVIS_KEEPALIVE_TIMEOUT     60000            /*!< How long to wait before retransmitting a provisional response (rfc 3261 13.3.1.1) */
#define MAX_AUTHTRIES                3                /*!< Try authentication three times, then fail */

#define SIP_MAX_HEADERS              64               /*!< Max amount of SIP headers to read */
#define SIP_MAX_LINES                256              /*!< Max amount of lines in SIP attachment (like SDP) */
#define SIP_MAX_PACKET               4096             /*!< Also from RFC 3261 (2543), should sub headers tho */

#define INITIAL_CSEQ                 101              /*!< our initial sip sequence number */

#define DEFAULT_MAX_SE               1800             /*!< Session-Timer Default Session-Expires period (RFC 4028) */
#define DEFAULT_MIN_SE               90               /*!< Session-Timer Default Min-SE period (RFC 4028) */

#define SDP_MAX_RTPMAP_CODECS        32               /*!< Maximum number of codecs allowed in received SDP */

/*! \brief Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
	.flags = 0,
	.max_size = -1,
	.resync_threshold = -1,
	.impl = "",
	.target_extra = -1,
};
static struct ast_jb_conf global_jbconf;	/*!< Global jitterbuffer configuration */

static const char config[] = "sip.conf";	/*!< Main configuration file */
static const char notify_config[] = "sip_notify.conf";	/*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */

#define RTP 	1
#define NO_RTP	0

/*! \brief Authorization scheme for call transfers 
\note Not a bitfield flag, since there are plans for other modes,
	like "only allow transfers for authenticated devices" */
enum transfermodes {
	TRANSFER_OPENFORALL,            /*!< Allow all SIP transfers */
	TRANSFER_CLOSED,                /*!< Allow no SIP transfers */
};


enum sip_result {
	AST_SUCCESS = 0,
	AST_FAILURE = -1,
};

/*! \brief States for the INVITE transaction, not the dialog 
	\note this is for the INVITE that sets up the dialog
*/
enum invitestates {
	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 xmittype {
	XMIT_CRITICAL = 2,              /*!< Transmit critical SIP message reliably, with re-transmits.
                                              If it fails, it's critical and will cause a teardown of the session */
	XMIT_RELIABLE = 1,              /*!< Transmit SIP message reliably, with re-transmits */
	XMIT_UNRELIABLE = 0,            /*!< Transmit SIP message without bothering with re-transmits */
};

enum parse_register_result {
	PARSE_REGISTER_DENIED,
	PARSE_REGISTER_FAILED,
	PARSE_REGISTER_UPDATE,
	PARSE_REGISTER_QUERY,
};

enum subscriptiontype { 
	NONE = 0,
	XPIDF_XML,
	DIALOG_INFO_XML,
	CPIM_PIDF_XML,
	PIDF_XML,
	MWI_NOTIFICATION
};

/*! \brief Subscription types that we support. We support
   - dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
   - SIMPLE presence used for device status
   - Voicemail notification subscriptions
*/
static const struct cfsubscription_types {
	enum subscriptiontype type;
	const char * const event;
	const char * const mediatype;
	const char * const text;
} subscription_types[] = {
	{ NONE,		   "-",        "unknown",	             "unknown" },
 	/* RFC 4235: SIP Dialog event package */
	{ DIALOG_INFO_XML, "dialog",   "application/dialog-info+xml", "dialog-info+xml" },
	{ CPIM_PIDF_XML,   "presence", "application/cpim-pidf+xml",   "cpim-pidf+xml" },  /* RFC 3863 */
	{ PIDF_XML,        "presence", "application/pidf+xml",        "pidf+xml" },       /* RFC 3863 */
	{ XPIDF_XML,       "presence", "application/xpidf+xml",       "xpidf+xml" },       /* Pre-RFC 3863 with MS additions */
	{ MWI_NOTIFICATION,	"message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
};


/*! \brief 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.
*/
enum sip_auth_type {
	PROXY_AUTH = 407,
	WWW_AUTH = 401,
};

/*! \brief Authentication result from check_auth* functions */
enum check_auth_result {
	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,
	AUTH_PEER_NOT_DYNAMIC = -6,
	AUTH_ACL_FAILED = -7,
	AUTH_BAD_TRANSPORT = -8,
};

/*! \brief States for outbound registrations (with register= lines in sip.conf */
enum sipregistrystate {
	REG_STATE_UNREGISTERED = 0,	/*!< We are not registered 
		 *  \note Initial state. We should have a timeout scheduled for the initial
		 * (or next) registration transmission, calling sip_reregister
		 */

	REG_STATE_REGSENT,	/*!< Registration request sent 
		 * \note sent initial request, waiting for an ack or a timeout to
		 * retransmit the initial request.
		*/

	REG_STATE_AUTHSENT,	/*!< We have tried to authenticate 
		 * \note entered after transmit_register with auth info,
		 * waiting for an ack.
		 */

	REG_STATE_REGISTERED,	/*!< Registered and done */

	REG_STATE_REJECTED,	/*!< Registration rejected *
		 * \note only used when the remote party has an expire larger than
		 * our max-expire. This is a final state from which we do not
		 * recover (not sure how correctly).
		 */

	REG_STATE_TIMEOUT,	/*!< Registration timed out *
		* \note XXX unused */

	REG_STATE_NOAUTH,	/*!< We have no accepted credentials
		 * \note fatal - no chance to proceed */

	REG_STATE_FAILED,	/*!< Registration failed after several tries
		 * \note fatal - no chance to proceed */
};

/*! \brief Modes in which Asterisk can be configured to run SIP Session-Timers */
enum st_mode {
        SESSION_TIMER_MODE_INVALID = 0, /*!< Invalid value */ 
        SESSION_TIMER_MODE_ACCEPT,      /*!< Honor inbound Session-Timer requests */
        SESSION_TIMER_MODE_ORIGINATE,   /*!< Originate outbound and honor inbound requests */
        SESSION_TIMER_MODE_REFUSE       /*!< Ignore inbound Session-Timers requests */
};

/*! \brief The entity playing the refresher role for Session-Timers */
enum st_refresher {
        SESSION_TIMER_REFRESHER_AUTO,    /*!< Negotiated                      */
        SESSION_TIMER_REFRESHER_UAC,     /*!< Session is refreshed by the UAC */
        SESSION_TIMER_REFRESHER_UAS      /*!< Session is refreshed by the UAS */
};

/*!< Define some SIP transports */
enum sip_transport {
	SIP_TRANSPORT_UDP = 1,
	SIP_TRANSPORT_TCP = 1 << 1,
	SIP_TRANSPORT_TLS = 1 << 2,
};

/*! \brief definition of a sip proxy server
 *
 * For outbound proxies, this is allocated in the SIP peer dynamically or
 * statically as the global_outboundproxy. The pointer in a SIP message is just
 * a pointer and should *not* be de-allocated.
 */
struct sip_proxy {
	char name[MAXHOSTNAMELEN];      /*!< DNS name of domain/host or IP */
	struct sockaddr_in ip;          /*!< Currently used IP address and port */
	time_t last_dnsupdate;          /*!< When this was resolved */
	enum sip_transport transport;
	int force;                      /*!< If it's an outbound proxy, Force use of this outbound proxy for all outbound requests */
	/* Room for a SRV record chain based on the name */
};

/*! \brief States whether a SIP message can create a dialog in Asterisk. */
enum can_create_dialog {
	CAN_NOT_CREATE_DIALOG,
	CAN_CREATE_DIALOG,
	CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
};

/*! \brief SIP Request methods known by Asterisk 

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

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

/*! \brief The core structure to setup dialogs. We parse incoming messages by using
	structure and then route the messages according to the type.

      \note Note that sip_methods[i].id == i must hold or the code breaks */
static const struct  cfsip_methods { 
	enum sipmethod id;
	int need_rtp;		/*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
	char * const text;
	enum can_create_dialog can_create;
} sip_methods[] = {
	{ SIP_UNKNOWN,	 RTP,    "-UNKNOWN-", 	CAN_CREATE_DIALOG },
	{ SIP_RESPONSE,	 NO_RTP, "SIP/2.0",	CAN_NOT_CREATE_DIALOG },
	{ SIP_REGISTER,	 NO_RTP, "REGISTER", 	CAN_CREATE_DIALOG },
 	{ SIP_OPTIONS,	 NO_RTP, "OPTIONS", 	CAN_CREATE_DIALOG },
	{ SIP_NOTIFY,	 NO_RTP, "NOTIFY", 	CAN_CREATE_DIALOG },
	{ SIP_INVITE,	 RTP,    "INVITE", 	CAN_CREATE_DIALOG },
	{ SIP_ACK,	 NO_RTP, "ACK", 	CAN_NOT_CREATE_DIALOG },
	{ SIP_PRACK,	 NO_RTP, "PRACK", 	CAN_NOT_CREATE_DIALOG },
	{ SIP_BYE,	 NO_RTP, "BYE", 	CAN_NOT_CREATE_DIALOG },
	{ SIP_REFER,	 NO_RTP, "REFER", 	CAN_CREATE_DIALOG },
	{ SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE", 	CAN_CREATE_DIALOG },
	{ SIP_MESSAGE,	 NO_RTP, "MESSAGE", 	CAN_CREATE_DIALOG },
	{ SIP_UPDATE,	 NO_RTP, "UPDATE", 	CAN_NOT_CREATE_DIALOG },
	{ SIP_INFO,	 NO_RTP, "INFO", 	CAN_NOT_CREATE_DIALOG },
	{ SIP_CANCEL,	 NO_RTP, "CANCEL", 	CAN_NOT_CREATE_DIALOG },
	{ SIP_PUBLISH,	 NO_RTP, "PUBLISH", 	CAN_CREATE_DIALOG_UNSUPPORTED_METHOD },
	{ SIP_PING,	 NO_RTP, "PING", 	CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
};

static unsigned int chan_idx;

/*!  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
*/
#define SUPPORTED		1
#define NOT_SUPPORTED		0

/* SIP options */
#define SIP_OPT_REPLACES	(1 << 0)
#define SIP_OPT_100REL		(1 << 1)
#define SIP_OPT_TIMER		(1 << 2)
#define SIP_OPT_EARLY_SESSION	(1 << 3)
#define SIP_OPT_JOIN		(1 << 4)
#define SIP_OPT_PATH		(1 << 5)
#define SIP_OPT_PREF		(1 << 6)
#define SIP_OPT_PRECONDITION	(1 << 7)
#define SIP_OPT_PRIVACY		(1 << 8)
#define SIP_OPT_SDP_ANAT	(1 << 9)
#define SIP_OPT_SEC_AGREE	(1 << 10)
#define SIP_OPT_EVENTLIST	(1 << 11)
#define SIP_OPT_GRUU		(1 << 12)
#define SIP_OPT_TARGET_DIALOG	(1 << 13)
#define SIP_OPT_NOREFERSUB	(1 << 14)
#define SIP_OPT_HISTINFO	(1 << 15)
#define SIP_OPT_RESPRIORITY	(1 << 16)
#define SIP_OPT_UNKNOWN		(1 << 17)


/*! \brief List of well-known SIP options. If we get this in a require,
   we should check the list and answer accordingly. */
static const struct cfsip_options {
	int id;			/*!< Bitmap ID */
	int supported;		/*!< Supported by Asterisk ? */
	char * const text;	/*!< Text id, as in standard */
} sip_options[] = {	/* XXX used in 3 places */
	/* RFC3891: Replaces: header for transfer */
	{ SIP_OPT_REPLACES,	SUPPORTED,	"replaces" },	
	/* One version of Polycom firmware has the wrong label */
	{ SIP_OPT_REPLACES,	SUPPORTED,	"replace" },	
	/* RFC3262: PRACK 100% reliability */
	{ SIP_OPT_100REL,	NOT_SUPPORTED,	"100rel" },	
	/* RFC4028: SIP Session-Timers */
	{ SIP_OPT_TIMER,	SUPPORTED,	"timer" },
	/* RFC3959: SIP Early session support */
	{ SIP_OPT_EARLY_SESSION, NOT_SUPPORTED,	"early-session" },
	/* RFC3911: SIP Join header support */
	{ SIP_OPT_JOIN,		NOT_SUPPORTED,	"join" },
	/* RFC3327: Path support */
	{ SIP_OPT_PATH,		NOT_SUPPORTED,	"path" },
	/* RFC3840: Callee preferences */
	{ SIP_OPT_PREF,		NOT_SUPPORTED,	"pref" },
	/* RFC3312: Precondition support */
	{ SIP_OPT_PRECONDITION,	NOT_SUPPORTED,	"precondition" },
	/* RFC3323: Privacy with proxies*/
	{ SIP_OPT_PRIVACY,	NOT_SUPPORTED,	"privacy" },
	/* RFC4092: Usage of the SDP ANAT Semantics in the SIP */
	{ SIP_OPT_SDP_ANAT,	NOT_SUPPORTED,	"sdp-anat" },
	/* RFC3329: Security agreement mechanism */
	{ SIP_OPT_SEC_AGREE,	NOT_SUPPORTED,	"sec_agree" },
	/* SIMPLE events:  RFC4662 */
	{ SIP_OPT_EVENTLIST,	NOT_SUPPORTED,	"eventlist" },
	/* GRUU: Globally Routable User Agent URI's */
	{ SIP_OPT_GRUU,		NOT_SUPPORTED,	"gruu" },
	/* RFC4538: Target-dialog */
	{ SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED,	"tdialog" },
	/* Disable the REFER subscription, RFC 4488 */
	{ SIP_OPT_NOREFERSUB,	NOT_SUPPORTED,	"norefersub" },
	/* ietf-sip-history-info-06.txt */
	{ SIP_OPT_HISTINFO,	NOT_SUPPORTED,	"histinfo" },
	/* ietf-sip-resource-priority-10.txt */
	{ SIP_OPT_RESPRIORITY,	NOT_SUPPORTED,	"resource-priority" },
};


/*! \brief SIP Methods we support 
	\todo This string should be set dynamically. We only support REFER and SUBSCRIBE is we have
	allowsubscribe and allowrefer on in sip.conf.
*/
#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO"

/*! \brief SIP Extensions we support */
#define SUPPORTED_EXTENSIONS "replaces, timer" 

/*! \brief Standard SIP and TLS port from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_SIP_PORT	5060
#define STANDARD_TLS_PORT	5061
/*! \note in many SIP headers, absence of a port number implies port 5060,
 * and this is why we cannot change the above constant.
 * There is a limited number of places in asterisk where we could,
 * in principle, use a different "default" port number, but
 * we do not support this feature at the moment.
 * You can run Asterisk with SIP on a different port with a configuration
 * option. If you change this value, the signalling will be incorrect.
 */

/*! \name DefaultValues Default values, set and reset in reload_config before reading configuration 

   These are default values in the source. There are other recommended values in the
   sip.conf.sample for new installations. These may differ to keep backwards compatibility,
   yet encouraging new behaviour on new installations 
 */
/*@{*/ 
#define DEFAULT_CONTEXT		"default"
#define DEFAULT_MOHINTERPRET    "default"
#define DEFAULT_MOHSUGGEST      ""
#define DEFAULT_VMEXTEN 	"asterisk"
#define DEFAULT_CALLERID 	"asterisk"
#define DEFAULT_NOTIFYMIME 	"application/simple-message-summary"
#define DEFAULT_ALLOWGUEST	TRUE
#define DEFAULT_CALLCOUNTER	FALSE
#define DEFAULT_SRVLOOKUP	TRUE		/*!< Recommended setting is ON */
#define DEFAULT_COMPACTHEADERS	FALSE
#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. */
#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. */
#define DEFAULT_TOS_VIDEO       0               /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_TEXT        0               /*!< Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_COS_SIP         4
#define DEFAULT_COS_AUDIO       5
#define DEFAULT_COS_VIDEO       6
#define DEFAULT_COS_TEXT        5
#define DEFAULT_ALLOW_EXT_DOM	TRUE
#define DEFAULT_REALM		"asterisk"
#define DEFAULT_NOTIFYRINGING	TRUE
#define DEFAULT_PEDANTIC	FALSE
#define DEFAULT_AUTOCREATEPEER	FALSE
#define DEFAULT_QUALIFY		FALSE
#define DEFAULT_REGEXTENONQUALIFY FALSE
#define DEFAULT_T1MIN		100		/*!< 100 MS for minimal roundtrip time */
#define DEFAULT_MAX_CALL_BITRATE (384)		/*!< Max bitrate for video */
#ifndef DEFAULT_USERAGENT
#define DEFAULT_USERAGENT "Asterisk PBX"	/*!< Default Useragent: header unless re-defined in sip.conf */
#define DEFAULT_SDPSESSION "Asterisk PBX"	/*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
#define DEFAULT_SDPOWNER "root"		/*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
#endif
/*@}*/ 

/*! \name DefaultSettings
	Default setttings are used as a channel setting and as a default when
	configuring devices 
*/
/*@{*/ 
static char default_context[AST_MAX_CONTEXT];
static char default_subscribecontext[AST_MAX_CONTEXT];
static char default_language[MAX_LANGUAGE];
static char default_callerid[AST_MAX_EXTENSION];
static char default_fromdomain[AST_MAX_EXTENSION];
static char default_notifymime[AST_MAX_EXTENSION];
static int default_qualify;		/*!< Default Qualify= setting */
static char default_vmexten[AST_MAX_EXTENSION];
static char default_mohinterpret[MAX_MUSICCLASS];  /*!< Global setting for moh class to use when put on hold */
static char default_mohsuggest[MAX_MUSICCLASS];	   /*!< Global setting for moh class to suggest when putting 
                                                    *   a bridged channel on hold */
static int default_maxcallbitrate;	/*!< Maximum bitrate for call */
static struct ast_codec_pref default_prefs;		/*!< Default codec prefs */

/*! \brief a place to store all global settings for the sip channel driver */
struct sip_settings {
	int peer_rtupdate;	/*!< G: Update database with registration data for peer? */
	int rtsave_sysname;	/*!< G: Save system name at registration? */
	int ignore_regexpire;	/*!< G: Ignore expiration of peer  */
};

static struct sip_settings sip_cfg;
/*@}*/ 

/*! \name GlobalSettings
	Global settings apply to the channel (often settings you can change in the general section
	of sip.conf
*/
/*@{*/ 
static int global_directrtpsetup;	/*!< Enable support for Direct RTP setup (no re-invites) */
static int global_limitonpeers;		/*!< Match call limit on peers only */
static int global_rtautoclear;		/*!< Realtime ?? */
static int global_notifyringing;	/*!< Send notifications on ringing */
static int global_notifyhold;		/*!< Send notifications on hold */
static int global_alwaysauthreject;	/*!< Send 401 Unauthorized for all failing requests */
static int global_srvlookup;		/*!< SRV Lookup on or off. Default is on */
static int pedanticsipchecking;		/*!< Extra checking ?  Default off */
static int autocreatepeer;		/*!< Auto creation of peers at registration? Default off. */
static int global_match_auth_username;		/*!< Match auth username if available instead of From: Default off. */
static int global_relaxdtmf;		/*!< Relax DTMF */
static int global_prematuremediafilter;	/*!< Enable/disable premature frames in a call (causing 183 early media) */
static int global_relaxdtmf;			/*!< Relax DTMF */
static int global_rtptimeout;		/*!< Time out call if no RTP */
static int global_rtpholdtimeout;	/*!< Time out call if no RTP during hold */
static int global_rtpkeepalive;		/*!< Send RTP keepalives */
static int global_reg_timeout;	
static int global_regattempts_max;	/*!< Registration attempts before giving up */
static int global_shrinkcallerid;	/*!< enable or disable shrinking of caller id  */
static int global_allowguest;		/*!< allow unauthenticated users/peers to connect? */
static int global_callcounter;		/*!< Enable call counters for all devices. This is currently enabled by setting the peer
						call-limit to 999. When we remove the call-limit from the code, we can make it
						with just a boolean flag in the device structure */
static int global_allowsubscribe;	/*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE 
					    the global setting is in globals_flags[1] */
static unsigned int global_tos_sip;		/*!< IP type of service for SIP packets */
static unsigned int global_tos_audio;		/*!< IP type of service for audio RTP packets */
static unsigned int global_tos_video;		/*!< IP type of service for video RTP packets */
static unsigned int global_tos_text;		/*!< IP type of service for text RTP packets */
static unsigned int global_cos_sip;		/*!< 802.1p class of service for SIP packets */
static unsigned int global_cos_audio;		/*!< 802.1p class of service for audio RTP packets */
static unsigned int global_cos_video;		/*!< 802.1p class of service for video RTP packets */
static unsigned int global_cos_text;		/*!< 802.1p class of service for text RTP packets */
static int compactheaders;		/*!< send compact sip headers */
static int recordhistory;		/*!< Record SIP history. Off by default */
static int dumphistory;			/*!< Dump history to verbose before destroying SIP dialog */
static char global_realm[MAXHOSTNAMELEN]; 		/*!< Default realm */
static char global_regcontext[AST_MAX_CONTEXT];		/*!< Context for auto-extensions */
static char global_useragent[AST_MAX_EXTENSION];	/*!< Useragent for the SIP channel */
static char global_sdpsession[AST_MAX_EXTENSION];	/*!< SDP session name for the SIP channel */
static char global_sdpowner[AST_MAX_EXTENSION];	/*!< SDP owner name for the SIP channel */
static int allow_external_domains;	/*!< Accept calls to external SIP domains? */
static int global_callevents;		/*!< Whether we send manager events or not */
static int global_t1;			/*!< T1 time */
static int global_t1min;		/*!< T1 roundtrip time minimum */
static int global_timer_b;    /*!< Timer B - RFC 3261 Section 17.1.1.2 */
static int global_regextenonqualify;  /*!< Whether to add/remove regexten when qualifying peers */
static int global_autoframing;          /*!< Turn autoframing on or off. */
static enum transfermodes global_allowtransfer;	/*!< SIP Refer restriction scheme */
static struct sip_proxy global_outboundproxy;	/*!< Outbound proxy */
static int global_matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */
static int global_qualifyfreq; /*!< Qualify frequency */


/*! \brief Codecs that we support by default: */
static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
static enum st_mode global_st_mode;           /*!< Mode of operation for Session-Timers           */
static enum st_refresher global_st_refresher; /*!< Session-Timer refresher                        */
static int global_min_se;                     /*!< Lowest threshold for session refresh interval  */
static int global_max_se;                     /*!< Highest threshold for session refresh interval */

static int global_dynamic_exclude_static = 0;	/*!< Exclude static peers from contact registrations */
/*@}*/

/*! \brief Global list of addresses dynamic peers are not allowed to use */
static struct ast_ha *global_contact_ha = NULL;

/*! \name Object counters @{
 * \bug These counters are not handled in a thread-safe way ast_atomic_fetchadd_int()
 * should be used to modify these values. */
static int suserobjs = 0;                /*!< Static users */
static int ruserobjs = 0;                /*!< Realtime users */
static int speerobjs = 0;                /*!< Static peers */
static int rpeerobjs = 0;                /*!< Realtime peers */
static int apeerobjs = 0;                /*!< Autocreated peer objects */
static int regobjs = 0;                  /*!< Registry objects */
/* }@ */

static struct ast_flags global_flags[2] = {{0}};        /*!< global SIP_ flags */
static int global_t38_maxdatagram;			/*!< global T.38 FaxMaxDatagram override */

static char used_context[AST_MAX_CONTEXT]; /*!< name of automatically created context for unloading */


AST_MUTEX_DEFINE_STATIC(netlock);

/*! \brief 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(monlock);

AST_MUTEX_DEFINE_STATIC(sip_reload_lock);

/*! \brief This is the thread for the monitor which checks for input on the channels
   which are not currently in use.  */
static pthread_t monitor_thread = AST_PTHREADT_NULL;

static int sip_reloading = FALSE;                       /*!< Flag for avoiding multiple reloads at the same time */
static enum channelreloadreason sip_reloadreason;       /*!< Reason for last reload/load of configuration */

static struct sched_context *sched;     /*!< The scheduling context */
static struct io_context *io;           /*!< The IO context */
static int *sipsock_read_id;            /*!< ID of IO entry for sipsock FD */

#define DEC_CALL_LIMIT	0
#define INC_CALL_LIMIT	1
#define DEC_CALL_RINGING 2
#define INC_CALL_RINGING 3

/*!< The SIP socket definition */
struct sip_socket {
	enum sip_transport type;
	int fd;
	uint16_t port;
	struct ast_tcptls_session_instance *tcptls_session;	/* If tcp or tls, a socket manager */
};

/*! \brief sip_request: The data grabbed from the UDP socket
 *
 * \verbatim
 * Incoming messages: we first store the data from the socket in data[],
 * adding a trailing \0 to make string parsing routines happy.
 * Then call parse_request() and req.method = find_sip_method();
 * to initialize the other fields. The \r\n at the end of each line is   
 * replaced by \0, so that data[] is not a conforming SIP message anymore.
 * After this processing, rlPart1 is set to non-NULL to remember
 * that we can run get_header() on this kind of packet.
 *
 * parse_request() splits the first line as follows:
 * Requests have in the first line      method uri SIP/2.0
 *      rlPart1 = method; rlPart2 = uri;
 * Responses have in the first line     SIP/2.0 NNN description
 *      rlPart1 = SIP/2.0; rlPart2 = NNN + description;
 *
 * For outgoing packets, we initialize the fields with init_req() or init_resp()
 * (which fills the first line to "METHOD uri SIP/2.0" or "SIP/2.0 code text"),
 * and then fill the rest with add_header() and add_line().
 * The \r\n at the end of the line are still there, so the get_header()
 * and similar functions don't work on these packets. 
 * \endverbatim
 */
struct sip_request {
	char *rlPart1; 	        /*!< SIP Method Name or "SIP/2.0" protocol version */
	char *rlPart2; 	        /*!< The Request URI or Response Status */
	int len;                /*!< bytes used in data[], excluding trailing null terminator. Rarely used. */
	int headers;            /*!< # of SIP Headers */
	int method;             /*!< Method of this request */
	int lines;              /*!< Body Content */
	unsigned int sdp_start; /*!< the line number where the SDP begins */
	unsigned int sdp_count; /*!< the number of lines of SDP */
	char debug;		/*!< print extra debugging if non zero */
	char has_to_tag;	/*!< non-zero if packet has To: tag */
	char ignore;		/*!< if non-zero This is a re-transmit, ignore it */
	char *header[SIP_MAX_HEADERS];
	char *line[SIP_MAX_LINES];
	char data[SIP_MAX_PACKET];
	/* XXX Do we need to unref socket.ser when the request goes away? */
	struct sip_socket socket;	/*!< The socket used for this request */
	AST_LIST_ENTRY(sip_request) next;
};

/*! \brief structure used in transfers */
struct sip_dual {
	struct ast_channel *chan1;	/*!< First channel involved */
	struct ast_channel *chan2;	/*!< Second channel involved */
	struct sip_request req;		/*!< Request that caused the transfer (REFER) */
	int seqno;			/*!< Sequence number */
};

struct sip_pkt;

/*! \brief Parameters to the transmit_invite function */
struct sip_invite_param {
	int addsipheaders;		/*!< Add extra SIP headers */
	const char *uri_options;	/*!< URI options to add to the URI */
	const char *vxml_url;		/*!< VXML url for Cisco phones */
	char *auth;			/*!< Authentication */
	char *authheader;		/*!< Auth header */
	enum sip_auth_type auth_type;	/*!< Authentication type */
	const char *replaces;		/*!< Replaces header for call transfers */
	int transfer;			/*!< Flag - is this Invite part of a SIP transfer? (invite/replaces) */
};

/*! \brief Structure to save routing information for a SIP session */
struct sip_route {
	struct sip_route *next;
	char hop[0];
};

/*! \brief Modes for SIP domain handling in the PBX */
enum domain_mode {
	SIP_DOMAIN_AUTO,		/*!< This domain is auto-configured */
	SIP_DOMAIN_CONFIG,		/*!< This domain is from configuration */
};

/*! \brief Domain data structure. 
	\note In the future, we will connect this to a configuration tree specific
	for this domain
*/
struct domain {
	char domain[MAXHOSTNAMELEN];		/*!< SIP domain we are responsible for */
	char context[AST_MAX_EXTENSION];	/*!< Incoming context for this domain */
	enum domain_mode mode;			/*!< How did we find this domain? */
	AST_LIST_ENTRY(domain) list;		/*!< List mechanics */
};

static AST_LIST_HEAD_STATIC(domain_list, domain);	/*!< The SIP domain list */


/*! \brief sip_history: Structure for saving transactions within a SIP dialog */
struct sip_history {
	AST_LIST_ENTRY(sip_history) list;
	char event[0];	/* actually more, depending on needs */
};

AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */

/*! \brief sip_auth: Credentials for authentication to other SIP services */
struct sip_auth {
	char realm[AST_MAX_EXTENSION];  /*!< Realm in which these credentials are valid */
	char username[256];             /*!< Username */
	char secret[256];               /*!< Secret */
	char md5secret[256];            /*!< MD5Secret */
	struct sip_auth *next;          /*!< Next auth structure in list */
};

/*! \name SIPflags
	Various flags for the flags field in the pvt structure 
	Trying to sort these up (one or more of the following):
	D: Dialog
	P: Peer/user
	G: Global flag
	When flags are used by multiple structures, it is important that
	they have a common layout so it is easy to copy them.
*/
/*@{*/ 
#define SIP_OUTGOING		(1 << 0)	/*!< D: Direction of the last transaction in this dialog */
#define SIP_RINGING		(1 << 2)	/*!< D: Have sent 180 ringing */
#define SIP_PROGRESS_SENT	(1 << 3)	/*!< D: Have sent 183 message progress */
#define SIP_NEEDREINVITE	(1 << 4)	/*!< D: Do we need to send another reinvite? */
#define SIP_PENDINGBYE		(1 << 5)	/*!< D: Need to send bye after we ack? */
#define SIP_GOTREFER		(1 << 6)	/*!< D: Got a refer? */
#define SIP_CALL_LIMIT		(1 << 7)	/*!< D: Call limit enforced for this call */
#define SIP_INC_COUNT		(1 << 8)	/*!< D: Did this dialog increment the counter of in-use calls? */
#define SIP_INC_RINGING		(1 << 9)	/*!< D: Did this connection increment the counter of in-use calls? */
#define SIP_DEFER_BYE_ON_TRANSFER	(1 << 10)	/*!< D: Do not hangup at first ast_hangup */

#define SIP_PROMISCREDIR	(1 << 11)	/*!< DP: Promiscuous redirection */
#define SIP_TRUSTRPID		(1 << 12)	/*!< DP: Trust RPID headers? */
#define SIP_USEREQPHONE		(1 << 13)	/*!< DP: Add user=phone to numeric URI. Default off */
#define SIP_USECLIENTCODE	(1 << 14)	/*!< DP: Trust X-ClientCode info message */

/* DTMF flags - see str2dtmfmode() and dtmfmode2str() */
#define SIP_DTMF		(7 << 15)	/*!< DP: DTMF Support: five settings, uses three bits */
#define SIP_DTMF_RFC2833	(0 << 15)	/*!< DP: DTMF Support: RTP DTMF - "rfc2833" */
#define SIP_DTMF_INBAND		(1 << 15)	/*!< DP: DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
#define SIP_DTMF_INFO		(2 << 15)	/*!< DP: DTMF Support: SIP Info messages - "info" */
#define SIP_DTMF_AUTO		(3 << 15)	/*!< DP: DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
#define SIP_DTMF_SHORTINFO      (4 << 15)       /*!< DP: DTMF Support: SIP Info messages - "info" - short variant */

/* NAT settings - see nat2str() */
#define SIP_NAT			(3 << 18)	/*!< DP: four settings, uses two bits */
#define SIP_NAT_NEVER		(0 << 18)	/*!< DP: No nat support */
#define SIP_NAT_RFC3581		(1 << 18)	/*!< DP: NAT RFC3581 */
#define SIP_NAT_ROUTE		(2 << 18)	/*!< DP: NAT Only ROUTE */
#define SIP_NAT_ALWAYS		(3 << 18)	/*!< DP: NAT Both ROUTE and RFC3581 */

/* re-INVITE related settings */
#define SIP_REINVITE		(7 << 20)	/*!< DP: four settings, uses three bits */
#define SIP_REINVITE_NONE	(0 << 20)	/*!< DP: no reinvite allowed */
#define SIP_CAN_REINVITE	(1 << 20)	/*!< DP: allow peers to be reinvited to send media directly p2p */
#define SIP_CAN_REINVITE_NAT	(2 << 20)	/*!< DP: allow media reinvite when new peer is behind NAT */
#define SIP_REINVITE_UPDATE	(4 << 20)	/*!< DP: use UPDATE (RFC3311) when reinviting this peer */

/* "insecure" settings - see insecure2str() */
#define SIP_INSECURE		(3 << 23)	/*!< DP: three settings, uses two bits */
#define SIP_INSECURE_NONE	(0 << 23)	/*!< DP: secure mode */
#define SIP_INSECURE_PORT	(1 << 23)	/*!< DP: don't require matching port for incoming requests */
#define SIP_INSECURE_INVITE	(1 << 24)	/*!< DP: don't require authentication for incoming INVITEs */

/* Sending PROGRESS in-band settings */
#define SIP_PROG_INBAND		(3 << 25)	/*!< DP: three settings, uses two bits */
#define SIP_PROG_INBAND_NEVER	(0 << 25)
#define SIP_PROG_INBAND_NO	(1 << 25)
#define SIP_PROG_INBAND_YES	(2 << 25)

#define SIP_SENDRPID		(1 << 29)	/*!< DP: Remote Party-ID Support */
#define SIP_G726_NONSTANDARD	(1 << 31)	/*!< DP: Use non-standard packing for G726-32 data */

/*! \brief Flags to copy from peer/user to dialog */
#define SIP_FLAGS_TO_COPY \
	(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
	 SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \
	 SIP_USEREQPHONE | SIP_INSECURE)
/*@}*/ 

/*! \name SIPflags2
	a second page of flags (for flags[1] */
/*@{*/ 
/* realtime flags */
#define SIP_PAGE2_RTCACHEFRIENDS	(1 << 0)	/*!< GP: Should we keep RT objects in memory for extended time? */
#define SIP_PAGE2_RTAUTOCLEAR		(1 << 2)	/*!< GP: Should we clean memory from peers after expiry? */
#define SIP_PAGE2_HAVEPEERCONTEXT	(1 << 3)	/*< Are we associated with a configured peer context? */
/* Space for addition of other realtime flags in the future */
#define SIP_PAGE2_STATECHANGEQUEUE	(1 << 9)	/*!< D: Unsent state pending change exists */

#define SIP_PAGE2_RPORT_PRESENT         (1 << 10)       /*!< Was rport received in the Via header? */
#define SIP_PAGE2_VIDEOSUPPORT		(1 << 14)	/*!< DP: Video supported if offered? */
#define SIP_PAGE2_TEXTSUPPORT		(1 << 15)	/*!< GDP: Global text enable */
#define SIP_PAGE2_ALLOWSUBSCRIBE	(1 << 16)	/*!< GP: Allow subscriptions from this peer? */
#define SIP_PAGE2_ALLOWOVERLAP		(1 << 17)	/*!< DP: Allow overlap dialing ? */
#define SIP_PAGE2_SUBSCRIBEMWIONLY	(1 << 18)	/*!< GP: Only issue MWI notification if subscribed to */
#define SIP_PAGE2_IGNORESDPVERSION	(1 << 19)	/*!< GDP: Ignore the SDP session version number we receive and treat all sessions as new */

#define SIP_PAGE2_T38SUPPORT		        (3 << 20)	/*!< GDP: T.38 Fax Support */
#define SIP_PAGE2_T38SUPPORT_UDPTL	        (1 << 20)	/*!< GDP: T.38 Fax Support (no error correction) */
#define SIP_PAGE2_T38SUPPORT_UDPTL_FEC	        (2 << 20)	/*!< GDP: T.38 Fax Support (FEC error correction) */
#define SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY	(3 << 20)	/*!< GDP: T.38 Fax Support (redundancy error correction) */

#define SIP_PAGE2_CALL_ONHOLD		(3 << 23)	/*!< D: Call hold states: */
#define SIP_PAGE2_CALL_ONHOLD_ACTIVE    (1 << 23)       /*!< D: Active hold */
#define SIP_PAGE2_CALL_ONHOLD_ONEDIR	(2 << 23)	/*!< D: One directional hold */
#define SIP_PAGE2_CALL_ONHOLD_INACTIVE	(3 << 23)	/*!< D: Inactive hold */

#define SIP_PAGE2_RFC2833_COMPENSATE    (1 << 25)	/*!< DP: Compensate for buggy RFC2833 implementations */
#define SIP_PAGE2_BUGGY_MWI		(1 << 26)	/*!< DP: Buggy CISCO MWI fix */
#define SIP_PAGE2_DIALOG_ESTABLISHED    (1 << 27)       /*!< 29: Has a dialog been established? */
#define SIP_PAGE2_FAX_DETECT		(1 << 28)		/*!< DP: Fax Detection support */
#define SIP_PAGE2_REGISTERTRYING        (1 << 29)       /*!< DP: Send 100 Trying on REGISTER attempts */
#define SIP_PAGE2_UDPTL_DESTINATION     (1 << 30)       /*!< DP: Use source IP of RTP as destination if NAT is enabled */

#define SIP_PAGE2_FLAGS_TO_COPY \
	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
	SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
	 SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_UDPTL_DESTINATION | \
	 SIP_PAGE2_FAX_DETECT | SIP_PAGE2_HAVEPEERCONTEXT)

/*@}*/ 

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

static enum sip_debug_e sipdebug;

/*! \brief extra debugging for 'text' related events.
 * At thie moment this is set together with sip_debug_console.
 * It should either go away or be implemented properly.
 */
static int sipdebug_text;

/*! \brief T38 States for a call */
enum t38state {
	T38_DISABLED = 0,                /*!< Not enabled */
	T38_LOCAL_REINVITE,              /*!< Offered from local - REINVITE */
	T38_PEER_REINVITE,               /*!< Offered from peer - REINVITE */
	T38_ENABLED                      /*!< Negotiated (enabled) */
};

/*! \brief T.38 channel settings (at some point we need to make this alloc'ed */
struct t38properties {
	enum t38state state;		/*!< T.38 state */
	struct ast_control_t38_parameters our_parms;
	struct ast_control_t38_parameters their_parms;
};

/*! \brief Parameters to know status of transfer */
enum referstatus {
	REFER_IDLE,                    /*!< No REFER is in progress */
	REFER_SENT,                    /*!< Sent REFER to transferee */
	REFER_RECEIVED,                /*!< Received REFER from transferrer */
	REFER_CONFIRMED,               /*!< Refer confirmed with a 100 TRYING (unused) */
	REFER_ACCEPTED,                /*!< Accepted by transferee */
	REFER_RINGING,                 /*!< Target Ringing */
	REFER_200OK,                   /*!< Answered by transfer target */
	REFER_FAILED,                  /*!< REFER declined - go on */
	REFER_NOAUTH                   /*!< We had no auth for REFER */
};

/*! \brief generic struct to map between strings and integers.
 * Fill it with x-s pairs, terminate with an entry with s = NULL;
 * Then you can call map_x_s(...) to map an integer to a string,
 * and map_s_x() for the string -> integer mapping.
 */
struct _map_x_s {
	int x;
	const char *s;
};              

static const struct _map_x_s referstatusstrings[] = {
	{ REFER_IDLE,		"<none>" },
	{ REFER_SENT,		"Request sent" },
	{ REFER_RECEIVED,	"Request received" },
	{ REFER_CONFIRMED,	"Confirmed" },
	{ REFER_ACCEPTED,	"Accepted" },
	{ REFER_RINGING,	"Target ringing" },
	{ REFER_200OK,		"Done" },
	{ REFER_FAILED,		"Failed" },
	{ REFER_NOAUTH,		"Failed - auth failure" },
	{ -1,			NULL} /* terminator */
};

/*! \brief Structure to handle SIP transfers. Dynamically allocated when needed
	\note OEJ: Should be moved to string fields */
struct sip_refer {
	char refer_to[AST_MAX_EXTENSION];		/*!< Place to store REFER-TO extension */
	char refer_to_domain[AST_MAX_EXTENSION];	/*!< Place to store REFER-TO domain */
	char refer_to_urioption[AST_MAX_EXTENSION];	/*!< Place to store REFER-TO uri options */
	char refer_to_context[AST_MAX_EXTENSION];	/*!< Place to store REFER-TO context */
	char referred_by[AST_MAX_EXTENSION];		/*!< Place to store REFERRED-BY extension */
	char referred_by_name[AST_MAX_EXTENSION];	/*!< Place to store REFERRED-BY extension */
	char refer_contact[AST_MAX_EXTENSION];		/*!< Place to store Contact info from a REFER extension */
	char replaces_callid[SIPBUFSIZE];			/*!< Replace info: callid */
	char replaces_callid_totag[SIPBUFSIZE/2];		/*!< Replace info: to-tag */
	char replaces_callid_fromtag[SIPBUFSIZE/2];		/*!< Replace info: from-tag */
	struct sip_pvt *refer_call;			/*!< Call we are referring. This is just a reference to a
							 * dialog owned by someone else, so we should not destroy
							 * it when the sip_refer object goes.
							 */
	int attendedtransfer