sip.h

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2010, Digium, Inc.
 *
 * 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 chan_sip header file
 */

#ifndef _SIP_H
#define _SIP_H

#include "asterisk.h"

#include "asterisk/stringfields.h"
#include "asterisk/linkedlists.h"
#include "asterisk/strings.h"
#include "asterisk/tcptls.h"
#include "asterisk/test.h"
#include "asterisk/channel.h"
#include "asterisk/app.h"
#include "asterisk/astobj.h"

#ifndef FALSE
#define FALSE    0
#endif

#ifndef TRUE
#define TRUE     1
#endif

/* Arguments for find_peer */
#define FINDUSERS (1 << 0)
#define FINDPEERS (1 << 1)
#define FINDALLDEVICES (FINDUSERS | FINDPEERS)

#define	SIPBUFSIZE      512             /*!< Buffer size for many operations */

#define XMIT_ERROR      -2

#define SIP_RESERVED    ";/?:@&=+$,# "  /*!< Reserved characters in the username part of the URI */

#define DEFAULT_DEFAULT_EXPIRY       120
#define DEFAULT_MIN_EXPIRY           60
#define DEFAULT_MAX_EXPIRY           3600
#define DEFAULT_MWI_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 */

#define DEFAULT_QUALIFY_GAP   100
#define DEFAULT_QUALIFY_PEERS 1

#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 DEFAULT_TIMER_T1          500         /*!< SIP timer T1 (according to RFC 3261) */
#define SIP_TRANS_TIMEOUT         64 * DEFAULT_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             64     /*!< Max amount of lines in SIP attachment (like SDP) */
#define SIP_MIN_PACKET            4096   /*!< Initialize size of memory to allocate for packets */
#define MAX_HISTORY_ENTRIES		  50	 /*!< Max entires in the history list for a sip_pvt */

#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 */

#define RTP     1
#define NO_RTP  0

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

/*! 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_FROMCHANGE    (1 << 17)
#define SIP_OPT_RECLISTINV    (1 << 18)
#define SIP_OPT_RECLISTSUB    (1 << 19)
#define SIP_OPT_OUTBOUND      (1 << 20)
#define SIP_OPT_UNKNOWN       (1 << 21)

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

/*! \brief SIP Extensions we support
 *  \note This should be generated based on the previous array
 *  in combination with settings.
 *
 *  \todo We should not have "timer" if it's disabled in the configuration file.
 */
#define SUPPORTED_EXTENSIONS "replaces, timer"

/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_SIP_PORT	5060
/*! \brief Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS */
#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 in the source code, 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"  /*!< The default context for [general] section as well as devices */
#define DEFAULT_MOHINTERPRET   "default"  /*!< The default music class */
#define DEFAULT_MOHSUGGEST     ""
#define DEFAULT_VMEXTEN        "asterisk" /*!< Default voicemail extension */
#define DEFAULT_CALLERID       "asterisk" /*!< Default caller ID */
#define DEFAULT_MWI_FROM       ""
#define DEFAULT_NOTIFYMIME     "application/simple-message-summary"
#define DEFAULT_ALLOWGUEST     TRUE
#define DEFAULT_RTPKEEPALIVE   0      /*!< Default RTPkeepalive setting */
#define DEFAULT_CALLCOUNTER    FALSE   /*!< Do not enable call counters by default */
#define DEFAULT_SRVLOOKUP      TRUE    /*!< Recommended setting is ON */
#define DEFAULT_COMPACTHEADERS FALSE  /*!< Send compact (one-character) SIP headers. Default off */
#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      /*!< Level 2 class of service for SIP signalling */
#define DEFAULT_COS_AUDIO      5      /*!< Level 2 class of service for audio media  */
#define DEFAULT_COS_VIDEO      6      /*!< Level 2 class of service for video media */
#define DEFAULT_COS_TEXT       5      /*!< Level 2 class of service for text media (T.140) */
#define DEFAULT_ALLOW_EXT_DOM  TRUE    /*!< Allow external domains */
#define DEFAULT_REALM          "asterisk" /*!< Realm for HTTP digest authentication */
#define DEFAULT_DOMAINSASREALM FALSE    /*!< Use the domain option to guess the realm for registration and invite requests */
#define DEFAULT_NOTIFYRINGING  TRUE     /*!< Notify devicestate system on ringing state */
#define DEFAULT_NOTIFYCID      DISABLED	/*!< Include CID with ringing notifications */
#define DEFAULT_PEDANTIC       FALSE    /*!< Avoid following SIP standards for dialog matching */
#define DEFAULT_AUTOCREATEPEER FALSE    /*!< Don't create peers automagically */
#define	DEFAULT_MATCHEXTERNIPLOCALLY FALSE /*!< Match extern IP locally default setting */
#define DEFAULT_QUALIFY        FALSE    /*!< Don't monitor devices */
#define DEFAULT_CALLEVENTS     FALSE    /*!< Extra manager SIP call events */
#define DEFAULT_ALWAYSAUTHREJECT  FALSE /*!< Don't reject authentication requests always */
#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 */
#define DEFAULT_ENGINE     "asterisk"      /*!< Default RTP engine to use for sessions */
#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_TESTLAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
#endif
/*@}*/

/*! \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_OFFER_CC        (1 << 1) /*!< D: Offer CC on subsequent responses */
#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 */
#define SIP_NAT_FORCE_RPORT     (1 << 18) /*!< DP: Force rport even if not present in the request */
#define SIP_NAT_RPORT_PRESENT   (1 << 19) /*!< DP: rport was present in the request */

/* 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_DIRECT_MEDIA     (1 << 20) /*!< DP: allow peers to be reinvited to send media directly p2p */
#define SIP_DIRECT_MEDIA_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         (3 << 29) /*!< DP: Remote Party-ID Support */
#define SIP_SENDRPID_NO      (0 << 29)
#define SIP_SENDRPID_PAI     (1 << 29) /*!< Use "P-Asserted-Identity" for rpid */
#define SIP_SENDRPID_RPID    (2 << 29) /*!< Use "Remote-Party-ID" for rpid */
#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_FORCE_RPORT | 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 <<  1)    /*!< GP: Should we clean memory from peers after expiry? */
#define SIP_PAGE2_RPID_UPDATE			(1 <<  2)
#define SIP_PAGE2_Q850_REASON			(1 <<  3)    /*!< DP: Get/send cause code via Reason header */

#define SIP_PAGE2_SYMMETRICRTP			(1 <<  4)    /*!< GDP: Whether symmetric RTP is enabled or not */
#define SIP_PAGE2_STATECHANGEQUEUE		(1 <<  5)    /*!< D: Unsent state pending change exists */

#define SIP_PAGE2_CONNECTLINEUPDATE_PEND	(1 <<  6)
#define SIP_PAGE2_RPID_IMMEDIATE		(1 <<  7)
#define SIP_PAGE2_RPORT_PRESENT			(1 <<  8)   /*!< Was rport received in the Via header? */
#define SIP_PAGE2_PREFERRED_CODEC		(1 <<  9)   /*!< GDP: Only respond with single most preferred joint codec */
#define SIP_PAGE2_VIDEOSUPPORT			(1 << 10)   /*!< DP: Video supported if offered? */
#define SIP_PAGE2_TEXTSUPPORT			(1 << 11)   /*!< GDP: Global text enable */
#define SIP_PAGE2_ALLOWSUBSCRIBE		(1 << 12)   /*!< GP: Allow subscriptions from this peer? */
#define SIP_PAGE2_ALLOWOVERLAP			(1 << 13)   /*!< DP: Allow overlap dialing ? */
#define SIP_PAGE2_SUBSCRIBEMWIONLY		(1 << 14)   /*!< GP: Only issue MWI notification if subscribed to */
#define SIP_PAGE2_IGNORESDPVERSION		(1 << 15)   /*!< GDP: Ignore the SDP session version number we receive and treat all sessions as new */

#define SIP_PAGE2_T38SUPPORT			(3 << 16)    /*!< GDP: T.38 Fax Support */
#define SIP_PAGE2_T38SUPPORT_UDPTL		(1 << 16)    /*!< GDP: T.38 Fax Support (no error correction) */
#define SIP_PAGE2_T38SUPPORT_UDPTL_FEC		(2 << 16)    /*!< GDP: T.38 Fax Support (FEC error correction) */
#define SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY	(3 << 16)    /*!< GDP: T.38 Fax Support (redundancy error correction) */

#define SIP_PAGE2_CALL_ONHOLD			(3 << 18)  /*!< D: Call hold states: */
#define SIP_PAGE2_CALL_ONHOLD_ACTIVE		(1 << 18)  /*!< D: Active hold */
#define SIP_PAGE2_CALL_ONHOLD_ONEDIR		(2 << 18)  /*!< D: One directional hold */
#define SIP_PAGE2_CALL_ONHOLD_INACTIVE		(3 << 18)  /*!< D: Inactive hold */

#define SIP_PAGE2_RFC2833_COMPENSATE		(1 << 20)  /*!< DP: Compensate for buggy RFC2833 implementations */
#define SIP_PAGE2_BUGGY_MWI			(1 << 21)  /*!< DP: Buggy CISCO MWI fix */
#define SIP_PAGE2_DIALOG_ESTABLISHED		(1 << 22)  /*!< 29: Has a dialog been established? */

#define SIP_PAGE2_FAX_DETECT			(3 << 23)  /*!< DP: Fax Detection support */
#define SIP_PAGE2_FAX_DETECT_CNG		(1 << 23)  /*!< DP: Fax Detection support - detect CNG in audio */
#define SIP_PAGE2_FAX_DETECT_T38		(2 << 23)  /*!< DP: Fax Detection support - detect T.38 reinvite from peer */
#define SIP_PAGE2_FAX_DETECT_BOTH		(3 << 23)  /*!< DP: Fax Detection support - detect both */

#define SIP_PAGE2_REGISTERTRYING		(1 << 24)  /*!< DP: Send 100 Trying on REGISTER attempts */
#define SIP_PAGE2_UDPTL_DESTINATION		(1 << 25)  /*!< DP: Use source IP of RTP as destination if NAT is enabled */
#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS		(1 << 26)  /*!< DP: Always set up video, even if endpoints don't support it */
#define SIP_PAGE2_HAVEPEERCONTEXT	(1 << 27)	/*< Are we associated with a configured peer context? */

#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_FAX_DETECT | \
	SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC | \
	SIP_PAGE2_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP |\
	SIP_PAGE2_Q850_REASON | SIP_PAGE2_HAVEPEERCONTEXT)

/*@}*/

/*----------------------------------------------------------*/
/*----                    ENUMS                         ----*/
/*----------------------------------------------------------*/

/*! \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 */
};

/*! \brief The result of a lot of functions */
enum sip_result {
	AST_SUCCESS = 0,		/*!< FALSE means success, funny enough */
	AST_FAILURE = -1,		/*!< Failure code */
};

/*! \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 */
};

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

/*! \brief Results from the parse_register() function */
enum parse_register_result {
	PARSE_REGISTER_DENIED,
	PARSE_REGISTER_FAILED,
	PARSE_REGISTER_UPDATE,
	PARSE_REGISTER_QUERY,
};

/*! \brief Type of subscription, based on the packages we do support, see \ref subscription_types */
enum subscriptiontype {
	NONE = 0,
	XPIDF_XML,
	DIALOG_INFO_XML,
	CPIM_PIDF_XML,
	PIDF_XML,
	MWI_NOTIFICATION,
	CALL_COMPLETION,
};

/*! \brief The number of media types in enum \ref media_type below. */
#define OFFERED_MEDIA_COUNT	4

/*! \brief Media types generate different "dummy answers" for not accepting the offer of 
	a media stream. We need to add definitions for each RTP profile. Secure RTP is not
	the same as normal RTP and will require a new definition */
enum media_type {
	SDP_AUDIO,   /*!< RTP/AVP Audio */
	SDP_VIDEO,   /*!< RTP/AVP Video */
	SDP_IMAGE,   /*!< Image udptl, not TCP or RTP */
	SDP_TEXT,    /*!< RTP/AVP Realtime Text */
};

/*! \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,
	AUTH_RTP_FAILED = 9,
};

/*! \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 */
};

/*! \brief Define some implemented SIP transports
	\note Asterisk does not support SCTP or UDP/DTLS
*/
enum sip_transport {
	SIP_TRANSPORT_UDP = 1,         /*!< Unreliable transport for SIP, needs retransmissions */
	SIP_TRANSPORT_TCP = 1 << 1,    /*!< Reliable, but unsecure */
	SIP_TRANSPORT_TLS = 1 << 2,    /*!< TCP/TLS - reliable and secure transport for signalling */
};

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

/*! \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 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,
};

/*! \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 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 */
};

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

enum t38_action_flag {
	SDP_T38_NONE = 0, /*!< Do not modify T38 information at all */
	SDP_T38_INITIATE, /*!< Remote side has requested T38 with us */
	SDP_T38_ACCEPT,   /*!< Remote side accepted our T38 request */
};

enum sip_tcptls_alert {
	TCPTLS_ALERT_DATA,  /*!< \brief There is new data to be sent out */
	TCPTLS_ALERT_STOP,  /*!< \brief A request to stop the tcp_handler thread */
};


/*----------------------------------------------------------*/
/*----                    STRUCTS                       ----*/
/*----------------------------------------------------------*/

/*! \brief definition of a sip proxy server
 *
 * For outbound proxies, a sip_peer will contain a reference to a
 * dynamically allocated instance of a sip_proxy. A sip_pvt may also
 * contain a reference to a peer's outboundproxy, or it may contain
 * a reference to the sip_cfg.outboundproxy.
 */
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 argument for the 'show channels|subscriptions' callback. */
struct __show_chan_arg {
	int fd;
	int subscriptions;
	int numchans;   /* return value */
};

/*! \name GlobalSettings
	Global settings apply to the channel (often settings you can change in the general section
	of sip.conf
*/
/*@{*/
/*! \brief a place to store all global settings for the sip channel driver

	These are settings that will be possibly to apply on a group level later on.
	\note Do not add settings that only apply to the channel itself and can't
	      be applied to devices (trunks, services, phones)
*/
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  */
	int rtautoclear;            /*!< Realtime ?? */
	int directrtpsetup;         /*!< Enable support for Direct RTP setup (no re-invites) */
	int pedanticsipchecking;    /*!< Extra checking ?  Default off */
	int autocreatepeer;         /*!< Auto creation of peers at registration? Default off. */
	int srvlookup;              /*!< SRV Lookup on or off. Default is on */
	int allowguest;             /*!< allow unauthenticated peers to connect? */
	int alwaysauthreject;       /*!< Send 401 Unauthorized for all failing requests */
	int compactheaders;         /*!< send compact sip headers */
	int allow_external_domains; /*!< Accept calls to external SIP domains? */
	int callevents;             /*!< Whether we send manager events or not */
	int regextenonqualify;      /*!< Whether to add/remove regexten when qualifying peers */
	int matchexterniplocally;   /*!< Match externip/externhost setting against localnet setting */
	char regcontext[AST_MAX_CONTEXT];  /*!< Context for auto-extensions */
	unsigned int disallowed_methods;   /*!< methods that we should never try to use */
	int notifyringing;          /*!< Send notifications on ringing */
	int notifyhold;             /*!< Send notifications on hold */
	enum notifycid_setting notifycid;  /*!< Send CID with ringing notifications */
	enum transfermodes allowtransfer;  /*!< SIP Refer restriction scheme */
	int allowsubscribe;         /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
	                                 the global setting is in globals_flags[1] */
	char realm[MAXHOSTNAMELEN]; /*!< Default realm */
	int domainsasrealm;         /*!< Use domains lists as realms */
	struct sip_proxy outboundproxy; /*!< Outbound proxy */
	char default_context[AST_MAX_CONTEXT];
	char default_subscribecontext[AST_MAX_CONTEXT];
	struct ast_ha *contact_ha;  /*! \brief Global list of addresses dynamic peers are not allowed to use */
	format_t capability;        /*!< Supported codecs */
};

/*! \brief The SIP socket definition */
struct sip_socket {
	enum sip_transport type;  /*!< UDP, TCP or TLS */
	int fd;                   /*!< Filed descriptor, the actual socket */
	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 {
	ptrdiff_t rlPart1;      /*!< Offset of the SIP Method Name or "SIP/2.0" protocol version */
	ptrdiff_t rlPart2;      /*!< Offset of 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 */
	ptrdiff_t header[SIP_MAX_HEADERS]; /*!< Array of offsets into the request string of each SIP header*/
	ptrdiff_t line[SIP_MAX_LINES];     /*!< Array of offsets into the request string of each SDP line*/
	struct ast_str *data;	
	/* 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 given a sip_request and an offset, return the char * that resides there
 *
 * It used to be that rlPart1, rlPart2, and the header and line arrays were character
 * pointers. They are now offsets into the ast_str portion of the sip_request structure.
 * To avoid adding a bunch of redundant pointer arithmetic to the code, this macro is
 * provided to retrieve the string at a particular offset within the request's buffer
 */
#define REQ_OFFSET_TO_STR(req,offset) (ast_str_buffer((req)->data) + ((req)->offset))

/*! \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 */
};

/*! \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 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 */
};

/*! \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 */
};

/*! \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 */
};

/*! \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 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;
};

/*! \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;                         /*!< Attended or blind transfer? */
	int localtransfer;                            /*!< Transfer to local domain? */
	enum referstatus status;                      /*!< REFER status */
};

/*! \brief Struct to handle custom SIP notify requests. Dynamically allocated when needed */
struct sip_notify {
	struct ast_variable *headers;
	struct ast_str *content;
};

/*! \brief Structure that encapsulates all attributes related to running
 *   SIP Session-Timers feature on a per dialog basis.
 */
struct sip_st_dlg {
	int st_active;                     /*!< Session-Timers on/off */
	int st_interval;                   /*!< Session-Timers negotiated session refresh interval */
	int st_schedid;                    /*!< Session-Timers ast_sched scheduler id */
	enum st_refresher st_ref;          /*!< Session-Timers session refresher */
	int st_expirys;                    /*!< Session-Timers number of expirys */
	int st_active_peer_ua;             /*!< Session-Timers on/off in peer UA */
	int st_cached_min_se;              /*!< Session-Timers cached Min-SE */
	int st_cached_max_se;              /*!< Session-Timers cached Session-Expires */
	enum st_mode st_cached_mode;       /*!< Session-Timers cached M.O. */
	enum st_refresher st_cached_ref;   /*!< Session-Timers cached refresher */
	unsigned char quit_flag:1;         /*!< Stop trying to lock; just quit */
};


/*! \brief Structure that encapsulates all attributes related to configuration
 *   of SIP Session-Timers feature on a per user/peer basis.
 */
struct sip_st_cfg {
	enum st_mode st_mode_oper;    /*!< Mode of operation for Session-Timers           */
	enum st_refresher st_ref;     /*!< Session-Timer refresher                        */
	int st_min_se;                /*!< Lowest threshold for session refresh interval  */
	int st_max_se;                /*!< Highest threshold for session refresh interval */
};

/*! \brief Structure for remembering offered media in an INVITE, to make sure we reply
	to all media streams. In theory. In practise, we try our best. */
struct offered_media {
	int offered;
	char codecs[128];
};

/*! \brief Structure used for each SIP dialog, ie. a call, a registration, a subscribe.
 * Created and initialized by sip_alloc(), the descriptor goes into the list of
 * descriptors (dialoglist).
 */
struct sip_pvt {
	struct sip_pvt *next;                   /*!< Next dialog in chain */
	enum invitestates invitestate;          /*!< Track state of SIP_INVITEs */
	int method;                             /*!< SIP method that opened this dialog */
	AST_DECLARE_STRING_FIELDS(
		AST_STRING_FIELD(callid);       /*!< Global CallID */
		AST_STRING_FIELD(randdata);     /*!< Random data */
		AST_STRING_FIELD(accountcode);  /*!< Account code */
		AST_STRING_FIELD(realm);        /*!< Authorization realm */
		AST_STRING_FIELD(nonce);        /*!< Authorization nonce */
		AST_STRING_FIELD(opaque);       /*!< Opaque nonsense */
		AST_STRING_FIELD(qop);          /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
		AST_STRING_FIELD(domain);       /*!< Authorization domain */
		AST_STRING_FIELD(from);         /*!< The From: header */
		AST_STRING_FIELD(useragent);    /*!< User agent in SIP request */
		AST_STRING_FIELD(exten);        /*!< Extension where to start */
		AST_STRING_FIELD(context);      /*!< Context for this call */
		AST_STRING_FIELD(subscribecontext); /*!< Subscribecontext */
		AST_STRING_FIELD(subscribeuri); /*!< Subscribecontext */
		AST_STRING_FIELD(fromdomain);   /*!< Domain to show in the from field */
		AST_STRING_FIELD(fromuser);     /*!< User to show in the user field */
		AST_STRING_FIELD(fromname);     /*!< Name to show in the user field */
		AST_STRING_FIELD(tohost);       /*!< Host we should put in the "to" field */
		AST_STRING_FIELD(todnid);       /*!< DNID of this call (overrides host) */
		AST_STRING_FIELD(language);     /*!< Default language for this call */
		AST_STRING_FIELD(mohinterpret); /*!< MOH class to use when put on hold */
		AST_STRING_FIELD(mohsuggest);   /*!< MOH class to suggest when putting a peer on hold */
		AST_STRING_FIELD(rdnis);        /*!< Referring DNIS */
		AST_STRING_FIELD(redircause);   /*!< Referring cause */
		AST_STRING_FIELD(theirtag);     /*!< Their tag */
		AST_STRING_FIELD(username);     /*!< [user] name */
		AST_STRING_FIELD(peername);     /*!< [peer] name, not set if [user] */
		AST_STRING_FIELD(authname);     /*!< Who we use for authentication */
		AST_STRING_FIELD(uri);          /*!< Original requested URI */
		AST_STRING_FIELD(okcontacturi); /*!< URI from the 200 OK on INVITE */
		AST_STRING_FIELD(peersecret);   /*!< Password */
		AST_STRING_FIELD(peermd5secret);
		AST_STRING_FIELD(cid_num);      /*!< Caller*ID number */
		AST_STRING_FIELD(cid_name);     /*!< Caller*ID name */
		AST_STRING_FIELD(mwi_from);     /*!< Name to place in the From header in outgoing NOTIFY requests */
		AST_STRING_FIELD(fullcontact);  /*!< The Contact: that the UA registers with us */
		                                /* we only store the part in <brackets> in this field. */
		AST_STRING_FIELD(our_contact);  /*!< Our contact header */
		AST_STRING_FIELD(url);          /*!< URL to be sent with next message to peer */
		AST_STRING_FIELD(parkinglot);   /*!< Parkinglot */
		AST_STRING_FIELD(engine);       /*!< RTP engine to use */
		AST_STRING_FIELD(dialstring);   /*!< The dialstring used to call this SIP endpoint */
	);
	char via[128];                          /*!< Via: header */
	struct sip_socket socket;               /*!< The socket used for this dialog */
	unsigned int ocseq;                     /*!< Current outgoing seqno */
	unsigned int icseq;                     /*!< Current incoming seqno */
	ast_group_t callgroup;                  /*!< Call group */
	ast_group_t pickupgroup;                /*!< Pickup group */
	int lastinvite;                         /*!< Last Cseq of invite */
	struct ast_flags flags[2];              /*!< SIP_ flags */

	/* boolean flags that don't belong in flags */
	unsigned short do_history:1;          /*!< Set if we want to record history */
	unsigned short alreadygone:1;         /*!< already destroyed by our peer */
	unsigned short needdestroy:1;         /*!< need to be destroyed by the monitor thread */
	unsigned short outgoing_call:1;       /*!< this is an outgoing call */
	unsigned short answered_elsewhere:1;  /*!< This call is cancelled due to answer on another channel */
	unsigned short novideo:1;             /*!< Didn't get video in invite, don't offer */
	unsigned short notext:1;              /*!< Text not supported  (?) */
	unsigned short session_modify:1;      /*!< Session modification request true/false  */
	unsigned short route_persistent:1;    /*!< Is this the "real" route? */
	unsigned short autoframing:1;         /*!< Whether to use our local configuration for frame sizes (off)
	                                       *   or respect the other endpoint's request for frame sizes (on)
	                                       *   for incoming calls
	                                       */
	char tag[11];                     /*!< Our tag for this session */
	int timer_t1;                     /*!< SIP timer T1, ms rtt */
	int timer_b;                      /*!< SIP timer B, ms */
	unsigned int sipoptions;          /*!< Supported SIP options on the other end */
	unsigned int reqsipoptions;       /*!< Required SIP options on the other end */
	struct ast_codec_pref prefs;      /*!< codec prefs */
	format_t capability;              /*!< Special capability (codec) */
	format_t jointcapability;         /*!< Supported capability at both ends (codecs) */
	format_t peercapability;          /*!< Supported peer capability */
	format_t prefcodec;               /*!< Preferred codec (outbound only) */
	int noncodeccapability;	          /*!< DTMF RFC2833 telephony-event */
	int jointnoncodeccapability;      /*!< Joint Non codec capability */
	format_t redircodecs;             /*!< Redirect codecs */
	int maxcallbitrate;               /*!< Maximum Call Bitrate for Video Calls */	
	int t38_maxdatagram;              /*!< T.38 FaxMaxDatagram override */
	int request_queue_sched_id;       /*!< Scheduler ID of any scheduled action to process queued requests */
	int provisional_keepalive_sched_id;   /*!< Scheduler ID for provisional responses that need to be sent out to avoid cancellation */
	const char *last_provisional;         /*!< The last successfully transmitted provisonal response message */
	int authtries;                        /*!< Times we've tried to authenticate */
	struct sip_proxy *outboundproxy;      /*!< Outbound proxy for this dialog. Use ref_proxy to set this instead of setting it directly*/
	struct t38properties t38;             /*!< T38 settings */
	struct sockaddr_in udptlredirip;      /*!< Where our T.38 UDPTL should be going if not to us */
	struct ast_udptl *udptl;              /*!< T.38 UDPTL session */
	int callingpres;                      /*!< Calling presentation */
	int expiry;                         /*!< How long we take to expire */
	int sessionversion;                 /*!< SDP Session Version */
	int sessionid;                      /*!< SDP Session ID */
	long branch;                        /*!< The branch identifier of this session */
	long invite_branch;                 /*!< The branch used when we sent the initial INVITE */
	int64_t sessionversion_remote;      /*!< Remote UA's SDP Session Version */
	unsigned int portinuri:1;           /*!< Non zero if a port has been specified, will also disable srv lookups */
	struct sockaddr_in sa;              /*!< Our peer */
	struct sockaddr_in redirip;         /*!< Where our RTP should be going if not to us */
	struct sockaddr_in vredirip;        /*!< Where our Video RTP should be going if not to us */
	struct sockaddr_in tredirip;        /*!< Where our Text RTP should be going if not to us */
	time_t lastrtprx;                   /*!< Last RTP received */
	time_t lastrtptx;                   /*!< Last RTP sent */
	int rtptimeout;                     /*!< RTP timeout time */
	struct sockaddr_in recv;            /*!< Received as */
	struct sockaddr_in ourip;           /*!< Our IP (as seen from the outside) */