[asterisk/asterisk.git] / 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 SIP over TCP
 * \todo SIP over TLS
 * \todo Better support of forking
 * \todo VIA branch tag transaction checking
 * \todo Transaction support
 *
 * \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_request(), that parses a bit more.
 * if it's a response to an outbound request, it's sent to handle_response().
 * If it is a request, handle_request sends it to one of a list of functions
 * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
 * sipsock_read locks the ast_channel if it exists (an active call) and
 * unlocks it after we have processed the SIP message.
 *
 * A new INVITE is sent to handle_request_invite(), that will end up
 * starting a new channel in the PBX, the new channel after that executing
 * in a separate channel thread. This is an incoming "call".
 * When the call is answered, either by a bridged channel or the PBX itself
 * the sip_answer() function is called.
 *
 * The actual media - Video or Audio - is mostly handled by the RTP subsystem
 * in rtp.c 
 * 
 * \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
 */


#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#include <stdio.h>
#include <ctype.h>
#include <string.h>
#include <unistd.h>
#include <sys/socket.h>
#include <sys/ioctl.h>
#include <net/if.h>
#include <errno.h>
#include <stdlib.h>
#include <fcntl.h>
#include <netdb.h>
#include <signal.h>
#include <sys/signal.h>
#include <netinet/in.h>
#include <netinet/in_systm.h>
#include <arpa/inet.h>
#include <netinet/ip.h>
#include <regex.h>

#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/logger.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/options.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
#include "asterisk/linkedlists.h"
#include "asterisk/stringfields.h"
#include "asterisk/monitor.h"
#include "asterisk/localtime.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/compiler.h"
#include "asterisk/threadstorage.h"
#include "asterisk/translate.h"
#include "asterisk/version.h"

#ifndef FALSE
#define FALSE    0
#endif

#ifndef TRUE
#define TRUE     1
#endif

#define VIDEO_CODEC_MASK        0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
#ifndef IPTOS_MINCOST
#define IPTOS_MINCOST           0x02
#endif

/* #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;
static int expiry = DEFAULT_EXPIRY;

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

#define CALLERID_UNKNOWN        "Unknown"

#define DEFAULT_MAXMS                2000             /*!< Qualification: Must be faster than 2 seconds by default */
#define DEFAULT_FREQ_OK              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            32000            /*!< 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 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_MAX_PACKET               4096             /*!< Also from RFC 3261 (2543), should sub headers tho */

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

/*! \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 = ""
};
static struct ast_jb_conf global_jbconf;

static const char config[] = "sip.conf";
static const char notify_config[] = "sip_notify.conf";

#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 */
};

/* 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 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_FAILED,
        PARSE_REGISTER_UPDATE,
        PARSE_REGISTER_QUERY,
};

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

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 SIP Request methods known by Asterisk */
enum sipmethod {
        SIP_UNKNOWN,            /* Unknown response */
        SIP_RESPONSE,           /* Not request, response to outbound request */
        SIP_REGISTER,
        SIP_OPTIONS,
        SIP_NOTIFY,
        SIP_INVITE,
        SIP_ACK,
        SIP_PRACK,              /* Not supported at all */
        SIP_BYE,
        SIP_REFER,
        SIP_SUBSCRIBE,
        SIP_MESSAGE,
        SIP_UPDATE,             /* We can send UPDATE; but not accept it */
        SIP_INFO,
        SIP_CANCEL,
        SIP_PUBLISH,            /* Not supported at all */
        SIP_PING,               /* Not supported at all, no standard but still implemented out there */
};

/*! \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,
};

/*! \brief States for outbound registrations (with register= lines in sip.conf */
enum sipregistrystate {
        REG_STATE_UNREGISTERED = 0,     /*!< We are not registred */
        REG_STATE_REGSENT,      /*!< Registration request sent */
        REG_STATE_AUTHSENT,     /*!< We have tried to authenticate */
        REG_STATE_REGISTERED,   /*!< Registered and done */
        REG_STATE_REJECTED,     /*!< Registration rejected */
        REG_STATE_TIMEOUT,      /*!< Registration timed out */
        REG_STATE_NOAUTH,       /*!< We have no accepted credentials */
        REG_STATE_FAILED,       /*!< Registration failed after several tries */
};

enum can_create_dialog {
        CAN_NOT_CREATE_DIALOG,
        CAN_CREATE_DIALOG,
        CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
};

/*! XXX 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 }
};

/*!  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

#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)

/*! \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,        NOT_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 */
#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"

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

/*! \brief Standard SIP port from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_SIP_PORT       5060
/* 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.
 */

/* 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_MWITIME         10
#define DEFAULT_ALLOWGUEST      TRUE
#define DEFAULT_SRVLOOKUP       FALSE           /*!< 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_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_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 */
#endif


/* 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 */

/* Global settings only apply to the channel */
static int global_limitonpeers;         /*!< Match call limit on peers only */
static int global_rtautoclear;
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 off, RFC behavior 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_rtptimeout;           /*!< Time out call if no RTP */
static int global_rtpholdtimeout;
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_allowguest;           /*!< allow unauthenticated users/peers to connect? */
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 int global_mwitime;              /*!< Time between MWI checks for peers */
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 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 int allow_external_domains;      /*!< Accept calls to external SIP domains? */
static int global_callevents;           /*!< Whether we send manager events or not */
static int global_t1min;                /*!< T1 roundtrip time minimum */
static int global_autoframing;          /*!< Turn autoframing on or off. */
static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */

/*! \brief Codecs that we support by default: */
static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
static int noncodeccapability = AST_RTP_DTMF;

/* Object counters */
static int suserobjs = 0;                /*!< Static users */
static int ruserobjs = 0;                /*!< Realtime users */
static int speerobjs = 0;                /*!< Statis 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 */

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

/*! \brief sip_request: The data grabbed from the UDP socket */
struct sip_request {
        char *rlPart1;          /*!< SIP Method Name or "SIP/2.0" protocol version */
        char *rlPart2;          /*!< The Request URI or Response Status */
        int len;                /*!< Length */
        int headers;            /*!< # of SIP Headers */
        int method;             /*!< Method of this request */
        int lines;              /*!< Body Content */
        unsigned int flags;     /*!< SIP_PKT Flags for this packet */
        char *header[SIP_MAX_HEADERS];
        char *line[SIP_MAX_LINES];
        char data[SIP_MAX_PACKET];
        unsigned int sdp_start; /*!< the line number where the SDP begins */
        unsigned int sdp_end;   /*!< the line number where the SDP ends */
};

/*
 * A sip packet is stored into the data[] buffer, with the header followed
 * by an empty line and the body of the message.
 * On outgoing packets, data is accumulated in data[] with len reflecting
 * the next available byte, headers and lines count the number of lines
 * in both parts. There are no '\0' in data[0..len-1].
 *
 * On received packet, the input read from the socket is copied into data[],
 * len is set and the string is NUL-terminated. Then a parser fills up
 * the other fields -header[] and line[] to point to the lines of the
 * message, rlPart1 and rlPart2 parse the first lnie as below:
 *
 * Requests have in the first line      METHOD URI SIP/2.0
 *      rlPart1 = method; rlPart2 = uri;
 * Responses have in the first line     SIP/2.0 code description
 *      rlPart1 = SIP/2.0; rlPart2 = code + description;
 *
 */

/*! \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 */
};

/*--- Various flags for the flags field in the pvt structure */
#define SIP_ALREADYGONE         (1 << 0)        /*!< Whether or not we've already been destroyed by our peer */
#define SIP_NEEDDESTROY         (1 << 1)        /*!< if we need to be destroyed by the monitor thread */
#define SIP_NOVIDEO             (1 << 2)        /*!< Didn't get video in invite, don't offer */
#define SIP_RINGING             (1 << 3)        /*!< Have sent 180 ringing */
#define SIP_PROGRESS_SENT       (1 << 4)        /*!< Have sent 183 message progress */
#define SIP_NEEDREINVITE        (1 << 5)        /*!< Do we need to send another reinvite? */
#define SIP_PENDINGBYE          (1 << 6)        /*!< Need to send bye after we ack? */
#define SIP_GOTREFER            (1 << 7)        /*!< Got a refer? */
#define SIP_PROMISCREDIR        (1 << 8)        /*!< Promiscuous redirection */
#define SIP_TRUSTRPID           (1 << 9)        /*!< Trust RPID headers? */
#define SIP_USEREQPHONE         (1 << 10)       /*!< Add user=phone to numeric URI. Default off */
#define SIP_REALTIME            (1 << 11)       /*!< Flag for realtime users */
#define SIP_USECLIENTCODE       (1 << 12)       /*!< Trust X-ClientCode info message */
#define SIP_OUTGOING            (1 << 13)       /*!< Direction of the last transaction in this dialog */
#define SIP_FREE_BIT            (1 << 14)       /*!< ---- */
#define SIP_DEFER_BYE_ON_TRANSFER       (1 << 15)       /*!< Do not hangup at first ast_hangup */
#define SIP_DTMF                (3 << 16)       /*!< DTMF Support: four settings, uses two bits */
#define SIP_DTMF_RFC2833        (0 << 16)       /*!< DTMF Support: RTP DTMF - "rfc2833" */
#define SIP_DTMF_INBAND         (1 << 16)       /*!< DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
#define SIP_DTMF_INFO           (2 << 16)       /*!< DTMF Support: SIP Info messages - "info" */
#define SIP_DTMF_AUTO           (3 << 16)       /*!< DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
/* NAT settings */
#define SIP_NAT                 (3 << 18)       /*!< four settings, uses two bits */
#define SIP_NAT_NEVER           (0 << 18)       /*!< No nat support */
#define SIP_NAT_RFC3581         (1 << 18)       /*!< NAT RFC3581 */
#define SIP_NAT_ROUTE           (2 << 18)       /*!< NAT Only ROUTE */
#define SIP_NAT_ALWAYS          (3 << 18)       /*!< NAT Both ROUTE and RFC3581 */
/* re-INVITE related settings */
#define SIP_REINVITE            (7 << 20)       /*!< three bits used */
#define SIP_CAN_REINVITE        (1 << 20)       /*!< allow peers to be reinvited to send media directly p2p */
#define SIP_CAN_REINVITE_NAT    (2 << 20)       /*!< allow media reinvite when new peer is behind NAT */
#define SIP_REINVITE_UPDATE     (4 << 20)       /*!< use UPDATE (RFC3311) when reinviting this peer */
/* "insecure" settings */
#define SIP_INSECURE_PORT       (1 << 23)       /*!< don't require matching port for incoming requests */
#define SIP_INSECURE_INVITE     (1 << 24)       /*!< don't require authentication for incoming INVITEs */
/* Sending PROGRESS in-band settings */
#define SIP_PROG_INBAND         (3 << 25)       /*!< 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_NO_HISTORY          (1 << 27)       /*!< Suppress recording request/response history */
#define SIP_CALL_LIMIT          (1 << 28)       /*!< Call limit enforced for this call */
#define SIP_SENDRPID            (1 << 29)       /*!< Remote Party-ID Support */
#define SIP_INC_COUNT           (1 << 30)       /*!< Did this connection increment the counter of in-use calls? */
#define SIP_G726_NONSTANDARD    (1 << 31)       /*!< Use non-standard packing for G726-32 data */

#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_PORT | SIP_INSECURE_INVITE)

/*--- a new page of flags (for flags[1] */
/* realtime flags */
#define SIP_PAGE2_RTCACHEFRIENDS        (1 << 0)
#define SIP_PAGE2_RTUPDATE              (1 << 1)
#define SIP_PAGE2_RTAUTOCLEAR           (1 << 2)
#define SIP_PAGE2_RT_FROMCONTACT        (1 << 4)
#define SIP_PAGE2_RTSAVE_SYSNAME        (1 << 5)
/* Space for addition of other realtime flags in the future */
#define SIP_PAGE2_IGNOREREGEXPIRE       (1 << 10)
#define SIP_PAGE2_DEBUG                 (3 << 11)
#define SIP_PAGE2_DEBUG_CONFIG          (1 << 11)
#define SIP_PAGE2_DEBUG_CONSOLE         (1 << 12)
#define SIP_PAGE2_DYNAMIC               (1 << 13)       /*!< Dynamic Peers register with Asterisk */
#define SIP_PAGE2_SELFDESTRUCT          (1 << 14)       /*!< Automatic peers need to destruct themselves */
#define SIP_PAGE2_VIDEOSUPPORT          (1 << 15)
#define SIP_PAGE2_ALLOWSUBSCRIBE        (1 << 16)       /*!< Allow subscriptions from this peer? */
#define SIP_PAGE2_ALLOWOVERLAP          (1 << 17)       /*!< Allow overlap dialing ? */
#define SIP_PAGE2_SUBSCRIBEMWIONLY      (1 << 18)       /*!< Only issue MWI notification if subscribed to */
#define SIP_PAGE2_INC_RINGING           (1 << 19)       /*!< Did this connection increment the counter of in-use calls? */
#define SIP_PAGE2_T38SUPPORT            (7 << 20)       /*!< T38 Fax Passthrough Support */
#define SIP_PAGE2_T38SUPPORT_UDPTL      (1 << 20)       /*!< 20: T38 Fax Passthrough Support */
#define SIP_PAGE2_T38SUPPORT_RTP        (2 << 20)       /*!< 21: T38 Fax Passthrough Support (not implemented) */
#define SIP_PAGE2_T38SUPPORT_TCP        (4 << 20)       /*!< 22: T38 Fax Passthrough Support (not implemented) */
#define SIP_PAGE2_CALL_ONHOLD           (3 << 23)       /*!< Call states */
#define SIP_PAGE2_CALL_ONHOLD_ONEDIR    (1 << 23)       /*!< 23: One directional hold */
#define SIP_PAGE2_CALL_ONHOLD_INACTIVE  (1 << 24)       /*!< 24: Inactive  */
#define SIP_PAGE2_RFC2833_COMPENSATE    (1 << 25)       /*!< 25: ???? */
#define SIP_PAGE2_BUGGY_MWI             (1 << 26)       /*!< 26: Buggy CISCO MWI fix */

#define SIP_PAGE2_FLAGS_TO_COPY \
        (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
        SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI)

/* SIP packet flags */
#define SIP_PKT_DEBUG           (1 << 0)        /*!< Debug this packet */
#define SIP_PKT_WITH_TOTAG      (1 << 1)        /*!< This packet has a to-tag */
#define SIP_PKT_IGNORE          (1 << 2)        /*!< This is a re-transmit, ignore it */

/* T.38 set of flags */
#define T38FAX_FILL_BIT_REMOVAL         (1 << 0)        /*!< Default: 0 (unset)*/
#define T38FAX_TRANSCODING_MMR                  (1 << 1)        /*!< Default: 0 (unset)*/
#define T38FAX_TRANSCODING_JBIG         (1 << 2)        /*!< Default: 0 (unset)*/
/* Rate management */
#define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF   (0 << 3)
#define T38FAX_RATE_MANAGEMENT_LOCAL_TCF        (1 << 3)        /*!< Unset for transferredTCF (UDPTL), set for localTCF (TPKT) */
/* UDP Error correction */
#define T38FAX_UDP_EC_NONE                      (0 << 4)        /*!< two bits, if unset NO t38UDPEC field in T38 SDP*/
#define T38FAX_UDP_EC_FEC                       (1 << 4)        /*!< Set for t38UDPFEC */
#define T38FAX_UDP_EC_REDUNDANCY                (2 << 4)        /*!< Set for t38UDPRedundancy */
/* T38 Spec version */
#define T38FAX_VERSION                          (3 << 6)        /*!< two bits, 2 values so far, up to 4 values max */
#define T38FAX_VERSION_0                        (0 << 6)        /*!< Version 0 */
#define T38FAX_VERSION_1                        (1 << 6)        /*!< Version 1 */
/* Maximum Fax Rate */
#define T38FAX_RATE_2400                        (1 << 8)        /*!< 2400 bps t38FaxRate */
#define T38FAX_RATE_4800                        (1 << 9)        /*!< 4800 bps t38FaxRate */
#define T38FAX_RATE_7200                        (1 << 10)       /*!< 7200 bps t38FaxRate */
#define T38FAX_RATE_9600                        (1 << 11)       /*!< 9600 bps t38FaxRate */
#define T38FAX_RATE_12000                       (1 << 12)       /*!< 12000 bps t38FaxRate */
#define T38FAX_RATE_14400                       (1 << 13)       /*!< 14400 bps t38FaxRate */

/*!< This is default: NO MMR and JBIG transcoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;

#define sipdebug                ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)
#define sipdebug_config         ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)
#define sipdebug_console        ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)

/*! \brief T38 States for a call */
enum t38state {
        T38_DISABLED = 0,                /*!< Not enabled */
        T38_LOCAL_DIRECT,                /*!< Offered from local */
        T38_LOCAL_REINVITE,              /*!< Offered from local - REINVITE */
        T38_PEER_DIRECT,                 /*!< Offered from peer */
        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 {
        struct ast_flags t38support;    /*!< Flag for udptl, rtp or tcp support for this session */
        int capability;                 /*!< Our T38 capability */
        int peercapability;             /*!< Peers T38 capability */
        int jointcapability;            /*!< Supported T38 capability at both ends */
        enum t38state state;            /*!< T.38 state */
};

/*! \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 */
        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 */
};

static const struct c_referstatusstring {
        enum referstatus status;
        char *text;
} referstatusstrings[] = {
        { REFER_IDLE,           "<none>" },
        { REFER_SENT,           "Request sent" },
        { REFER_RECEIVED,       "Request received" },
        { REFER_ACCEPTED,       "Accepted" },
        { REFER_RINGING,        "Target ringing" },
        { REFER_200OK,          "Done" },
        { REFER_FAILED,         "Failed" },
        { REFER_NOAUTH,         "Failed - auth failure" }
} ;

/*! \brief Structure to handle SIP transfers. Dynamically allocated when needed  */
/* 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[BUFSIZ];                   /*!< Replace info: callid */
        char replaces_callid_totag[BUFSIZ/2];           /*!< Replace info: to-tag */
        char replaces_callid_fromtag[BUFSIZ/2];         /*!< Replace info: from-tag */
        struct sip_pvt *refer_call;                     /*!< Call we are referring */
        int attendedtransfer;                           /*!< Attended or blind transfer? */
        int localtransfer;                              /*!< Transfer to local domain? */
        enum referstatus status;                        /*!< REFER status */
};

/*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe  */
struct sip_pvt {
        ast_mutex_t pvt_lock;                   /*!< Dialog private lock */
        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(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(via);          /*!< Via: header */
                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(rpid);         /*!< Our RPID header */
                AST_STRING_FIELD(rpid_from);    /*!< Our RPID From header */
        );
        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 */
        int timer_t1;                           /*!< SIP timer T1, ms rtt */
        unsigned int sipoptions;                /*!< Supported SIP options on the other end */
        struct ast_codec_pref prefs;            /*!< codec prefs */
        int capability;                         /*!< Special capability (codec) */
        int jointcapability;                    /*!< Supported capability at both ends (codecs) */
        int peercapability;                     /*!< Supported peer capability */
        int prefcodec;                          /*!< Preferred codec (outbound only) */
        int noncodeccapability;                 /*!< DTMF RFC2833 telephony-event */
        int redircodecs;                        /*!< Redirect codecs */
        int maxcallbitrate;                     /*!< Maximum Call Bitrate for Video Calls */    
        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 authtries;                          /*!< Times we've tried to authenticate */
        int expiry;                             /*!< How long we take to expire */
        long branch;                            /*!< The branch identifier of this session */
        char tag[11];                           /*!< Our tag for this session */
        int sessionid;                          /*!< SDP Session ID */
        int sessionversion;                     /*!< SDP Session Version */
        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 */
        time_t lastrtprx;                       /*!< Last RTP received */
        time_t lastrtptx;                       /*!< Last RTP sent */
        int rtptimeout;                         /*!< RTP timeout time */
        struct sockaddr_in recv;                /*!< Received as */
        struct in_addr ourip;                   /*!< Our IP */
        struct ast_channel *owner;              /*!< Who owns us (if we have an owner) */
        struct sip_route *route;                /*!< Head of linked list of routing steps (fm Record-Route) */
        int route_persistant;                   /*!< Is this the "real" route? */
        struct sip_auth *peerauth;              /*!< Realm authentication */
        int noncecount;                         /*!< Nonce-count */
        char lastmsg[256];                      /*!< Last Message sent/received */
        int amaflags;                           /*!< AMA Flags */
        int pendinginvite;                      /*!< Any pending invite ? (seqno of this) */
        struct sip_request initreq;             /*!< Latest request that opened a new transaction
                                                        within this dialog.
                                                        NOT the request that opened the dialog
                                                */
        
        int initid;                             /*!< Auto-congest ID if appropriate (scheduler) */
        int autokillid;                         /*!< Auto-kill ID (scheduler) */
        enum transfermodes allowtransfer;       /*!< REFER: restriction scheme */
        struct sip_refer *refer;                /*!< REFER: SIP transfer data structure */
        enum subscriptiontype subscribed;       /*!< SUBSCRIBE: Is this dialog a subscription?  */
        int stateid;                            /*!< SUBSCRIBE: ID for devicestate subscriptions */
        int laststate;                          /*!< SUBSCRIBE: Last known extension state */
        int dialogver;                          /*!< SUBSCRIBE: Version for subscription dialog-info */
        
        struct ast_dsp *vad;                    /*!< Inband DTMF Detection dsp */
        
        struct sip_peer *relatedpeer;           /*!< If this dialog is related to a peer, which one 
                                                        Used in peerpoke, mwi subscriptions */
        struct sip_registry *registry;          /*!< If this is a REGISTER dialog, to which registry */
        struct ast_rtp *rtp;                    /*!< RTP Session */
        struct ast_rtp *vrtp;                   /*!< Video RTP session */
        struct sip_pkt *packets;                /*!< Packets scheduled for re-transmission */
        struct sip_history_head *history;       /*!< History of this SIP dialog */
        struct ast_variable *chanvars;          /*!< Channel variables to set for inbound call */
        struct sip_pvt *next;                   /*!< Next dialog in chain */
        struct sip_invite_param *options;       /*!< Options for INVITE */
        int autoframing;                        /*!< The number of Asters we group in a Pyroflax
                                                        before strolling to the Grokyzpå
                                                        (A bit unsure of this, please correct if
                                                        you know more) */
};

static struct sip_pvt *dialoglist = NULL;

/*! \brief Protect the SIP dialog list (of sip_pvt's) */
AST_MUTEX_DEFINE_STATIC(dialoglock);

/*! \brief hide the way the list is locked/unlocked */
static void dialoglist_lock(void)
{
        ast_mutex_lock(&dialoglock);
}

static void dialoglist_unlock(void)
{
        ast_mutex_unlock(&dialoglock);
}

#define FLAG_RESPONSE (1 << 0)
#define FLAG_FATAL (1 << 1)

/*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission */
struct sip_pkt {
        struct sip_pkt *next;                   /*!< Next packet in linked list */
        int retrans;                            /*!< Retransmission number */
        int method;                             /*!< SIP method for this packet */
        int seqno;                              /*!< Sequence number */
        unsigned int flags;                     /*!< non-zero if this is a response packet (e.g. 200 OK) */
        struct sip_pvt *owner;                  /*!< Owner AST call */
        int retransid;                          /*!< Retransmission ID */
        int timer_a;                            /*!< SIP timer A, retransmission timer */
        int timer_t1;                           /*!< SIP Timer T1, estimated RTT or 500 ms */
        int packetlen;                          /*!< Length of packet */
        char data[0];
};      

/*! \brief Structure for SIP user data. User's place calls to us */
struct sip_user {
        /* Users who can access various contexts */
        ASTOBJ_COMPONENTS(struct sip_user);
        char secret[80];                /*!< Password */
        char md5secret[80];             /*!< Password in md5 */
        char context[AST_MAX_CONTEXT];  /*!< Default context for incoming calls */
        char subscribecontext[AST_MAX_CONTEXT]; /* Default context for subscriptions */
        char cid_num[80];               /*!< Caller ID num */
        char cid_name[80];              /*!< Caller ID name */
        char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */
        char language[MAX_LANGUAGE];    /*!< Default language for this user */
        char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
        char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
        char useragent[256];            /*!< User agent in SIP request */
        struct ast_codec_pref prefs;    /*!< codec prefs */
        ast_group_t callgroup;          /*!< Call group */
        ast_group_t pickupgroup;        /*!< Pickup Group */
        unsigned int sipoptions;        /*!< Supported SIP options */
        struct ast_flags flags[2];      /*!< SIP_ flags */
        int amaflags;                   /*!< AMA flags for billing */
        int callingpres;                /*!< Calling id presentation */
        int capability;                 /*!< Codec capability */
        int inUse;                      /*!< Number of calls in use */
        int call_limit;                 /*!< Limit of concurrent calls */
        enum transfermodes allowtransfer;       /*! SIP Refer restriction scheme */
        struct ast_ha *ha;              /*!< ACL setting */
        struct ast_variable *chanvars;  /*!< Variables to set for channel created by user */
        int maxcallbitrate;             /*!< Maximum Bitrate for a video call */
        int autoframing;
};

/*! \brief Structure for SIP peer data, we place calls to peers if registered  or fixed IP address (host) */
/* XXX field 'name' must be first otherwise sip_addrcmp() will fail */
struct sip_peer {
        ASTOBJ_COMPONENTS(struct sip_peer);     /*!< name, refcount, objflags,  object pointers */
                                        /*!< peer->name is the unique name of this object */
        char secret[80];                /*!< Password */
        char md5secret[80];             /*!< Password in MD5 */
        struct sip_auth *auth;          /*!< Realm authentication list */
        char context[AST_MAX_CONTEXT];  /*!< Default context for incoming calls */
        char subscribecontext[AST_MAX_CONTEXT]; /*!< Default context for subscriptions */
        char username[80];              /*!< Temporary username until registration */ 
        char accountcode[AST_MAX_ACCOUNT_CODE]; /*!< Account code */
        int amaflags;                   /*!< AMA Flags (for billing) */
        char tohost[MAXHOSTNAMELEN];    /*!< If not dynamic, IP address */
        char regexten[AST_MAX_EXTENSION]; /*!< Extension to register (if regcontext is used) */
        char fromuser[80];              /*!< From: user when calling this peer */
        char fromdomain[MAXHOSTNAMELEN];        /*!< From: domain when calling this peer */
        char fullcontact[256];          /*!< Contact registered with us (not in sip.conf) */
        char cid_num[80];               /*!< Caller ID num */
        char cid_name[80];              /*!< Caller ID name */
        int callingpres;                /*!< Calling id presentation */
        int inUse;                      /*!< Number of calls in use */
        int inRinging;                  /*!< Number of calls ringing */
        int onHold;                     /*!< Peer has someone on hold */
        int call_limit;                 /*!< Limit of concurrent calls */
        enum transfermodes allowtransfer;       /*! SIP Refer restriction scheme */
        char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
        char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
        char language[MAX_LANGUAGE];    /*!<  Default language for prompts */
        char mohinterpret[MAX_MUSICCLASS];/*!<  Music on Hold class */
        char mohsuggest[MAX_MUSICCLASS];/*!<  Music on Hold class */
        char useragent[256];            /*!<  User agent in SIP request (saved from registration) */
        struct ast_codec_pref prefs;    /*!<  codec prefs */
        int lastmsgssent;
        time_t  lastmsgcheck;           /*!<  Last time we checked for MWI */
        unsigned int sipoptions;        /*!<  Supported SIP options */
        struct ast_flags flags[2];      /*!<  SIP_ flags */
        int expire;                     /*!<  When to expire this peer registration */
        int capability;                 /*!<  Codec capability */
        int rtptimeout;                 /*!<  RTP timeout */
        int rtpholdtimeout;             /*!<  RTP Hold Timeout */
        int rtpkeepalive;               /*!<  Send RTP packets for keepalive */
        ast_group_t callgroup;          /*!<  Call group */
        ast_group_t pickupgroup;        /*!<  Pickup group */
        struct ast_dnsmgr_entry *dnsmgr;/*!<  DNS refresh manager for peer */
        struct sockaddr_in addr;        /*!<  IP address of peer */
        int maxcallbitrate;             /*!< Maximum Bitrate for a video call */
        
        /* Qualification */
        struct sip_pvt *call;           /*!<  Call pointer */
        int pokeexpire;                 /*!<  When to expire poke (qualify= checking) */
        int lastms;                     /*!<  How long last response took (in ms), or -1 for no response */
        int maxms;                      /*!<  Max ms we will accept for the host to be up, 0 to not monitor */
        struct timeval ps;              /*!<  Time for sending SIP OPTION in sip_pke_peer() */
        struct sockaddr_in defaddr;     /*!<  Default IP address, used until registration */
        struct ast_ha *ha;              /*!<  Access control list */
        struct ast_variable *chanvars;  /*!<  Variables to set for channel created by user */
        struct sip_pvt *mwipvt;         /*!<  Subscription for MWI */
        int lastmsg;
        int autoframing;
};



/*! \brief Registrations with other SIP proxies */
struct sip_registry {
        ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1);
        AST_DECLARE_STRING_FIELDS(
                AST_STRING_FIELD(callid);       /*!< Global Call-ID */
                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(username);     /*!< Who we are registering as */
                AST_STRING_FIELD(authuser);     /*!< Who we *authenticate* as */
                AST_STRING_FIELD(hostname);     /*!< Domain or host we register to */
                AST_STRING_FIELD(secret);       /*!< Password in clear text */  
                AST_STRING_FIELD(md5secret);    /*!< Password in md5 */
                AST_STRING_FIELD(callback);     /*!< Contact extension */
                AST_STRING_FIELD(random);
        );
        int portno;                     /*!<  Optional port override */
        int expire;                     /*!< Sched ID of expiration */
        int expiry;                     /*!< Value to use for the Expires header */
        int regattempts;                /*!< Number of attempts (since the last success) */
        int timeout;                    /*!< sched id of sip_reg_timeout */
        int refresh;                    /*!< How often to refresh */
        struct sip_pvt *call;           /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */
        enum sipregistrystate regstate; /*!< Registration state (see above) */
        time_t regtime;         /*!< Last successful registration time */
        int callid_valid;               /*!< 0 means we haven't chosen callid for this registry yet. */
        unsigned int ocseq;             /*!< Sequence number we got to for REGISTERs for this registry */
        struct sockaddr_in us;          /*!< Who the server thinks we are */
        int noncecount;                 /*!< Nonce-count */
        char lastmsg[256];              /*!< Last Message sent/received */
};

/* --- Linked lists of various objects --------*/

/*! \brief  The user list: Users and friends */
static struct ast_user_list {
        ASTOBJ_CONTAINER_COMPONENTS(struct sip_user);
} userl;

/*! \brief  The peer list: Peers and Friends */
static struct ast_peer_list {
        ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer);
} peerl;

/*! \brief  The register list: Other SIP proxies we register with and place calls to */
static struct ast_register_list {
        ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
        int recheck;
} regl;

static int temp_pvt_init(void *);
static void temp_pvt_cleanup(void *);

/*! \brief A per-thread temporary pvt structure */
AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);

/*! \todo Move the sip_auth list to AST_LIST */
static struct sip_auth *authl = NULL;           /*!< Authentication list for realm authentication */


/* --- Sockets and networking --------------*/
static int sipsock  = -1;                       /*!< Main socket for SIP network communication */
static struct sockaddr_in bindaddr = { 0, };    /*!< The address we bind to */
static struct sockaddr_in externip;             /*!< External IP address if we are behind NAT */
static char externhost[MAXHOSTNAMELEN];         /*!< External host name (possibly with dynamic DNS and DHCP */
static time_t externexpire = 0;                 /*!< Expiration counter for re-resolving external host name in dynamic DNS */
static int externrefresh = 10;
static struct ast_ha *localaddr;                /*!< List of local networks, on the same side of NAT as this Asterisk */
static struct in_addr __ourip;
static struct sockaddr_in outboundproxyip;
static int ourport;
static struct sockaddr_in debugaddr;

static struct ast_config *notify_types;         /*!< The list of manual NOTIFY types we know how to send */

/*---------------------------- Forward declarations of functions in chan_sip.c */
/*! \note This is added to help splitting up chan_sip.c into several files
        in coming releases */

/*--- PBX interface functions */
static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause);
static int sip_devicestate(void *data);
static int sip_sendtext(struct ast_channel *ast, const char *text);
static int sip_call(struct ast_channel *ast, char *dest, int timeout);
static int sip_hangup(struct ast_channel *ast);
static int sip_answer(struct ast_channel *ast);
static struct ast_frame *sip_read(struct ast_channel *ast);
static int sip_write(struct ast_channel *ast, struct ast_frame *frame);
static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
static int sip_transfer(struct ast_channel *ast, const char *dest);
static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static int sip_senddigit_begin(struct ast_channel *ast, char digit);
static int sip_senddigit_end(struct ast_channel *ast, char digit);

/*--- Transmitting responses and requests */
static int sipsock_read(int *id, int fd, short events, void *ignore);
static int __sip_xmit(struct sip_pvt *p, char *data, int len);
static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod);
static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
static int retrans_pkt(void *data);
static int transmit_sip_request(struct sip_pvt *p, struct sip_request *req);
static int transmit_response_using_temp(ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg);
static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale);
static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, int reliable);
static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch);
static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init);
static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version);
static int transmit_info_with_digit(struct sip_pvt *p, const char digit);
static int transmit_info_with_vidupdate(struct sip_pvt *p);
static int transmit_message_with_text(struct sip_pvt *p, const char *text);
static int transmit_refer(struct sip_pvt *p, const char *dest);
static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, char *vmexten);
static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
static void copy_request(struct sip_request *dst, const struct sip_request *src);
static void receive_message(struct sip_pvt *p, struct sip_request *req);
static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req);
static int sip_send_mwi_to_peer(struct sip_peer *peer);
static int does_peer_need_mwi(struct sip_peer *peer);

/*--- Dialog management */
static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
                                 int useglobal_nat, const int intended_method);
static int __sip_autodestruct(void *data);
static void sip_scheddestroy(struct sip_pvt *p, int ms);
static void sip_cancel_destroy(struct sip_pvt *p);
static void sip_destroy(struct sip_pvt *p);
static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist);
static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
static void __sip_pretend_ack(struct sip_pvt *p);
static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
static int auto_congest(void *nothing);
static int update_call_counter(struct sip_pvt *fup, int event);
static int hangup_sip2cause(int cause);
static const char *hangup_cause2sip(int cause);
static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method);
static void free_old_route(struct sip_route *route);
static void list_route(struct sip_route *route);
static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards);
static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr_in *sin,
                                              struct sip_request *req, char *uri);
static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag);
static void check_pendings(struct sip_pvt *p);
static void *sip_park_thread(void *stuff);
static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno);
static int sip_sipredirect(struct sip_pvt *p, const char *dest);

/*--- Codec handling / SDP */
static void try_suggested_sip_codec(struct sip_pvt *p);
static const char* get_sdp_iterate(int* start, struct sip_request *req, const char *name);
static const char *get_sdp(struct sip_request *req, const char *name);
static int find_sdp(struct sip_request *req);
static int process_sdp(struct sip_pvt *p, struct sip_request *req);
static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
                             char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
                             int debug, int *min_packet_size);
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
                                char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
                                int debug);
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p);
static void do_setnat(struct sip_pvt *p, int natflags);

/*--- Authentication stuff */
static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
                                         const char *secret, const char *md5secret, int sipmethod,
                                         char *uri, enum xmittype reliable, int ignore);
static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
                                              int sipmethod, char *uri, enum xmittype reliable,
                                              struct sockaddr_in *sin, struct sip_peer **authpeer);
static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin);

/*--- Domain handling */
static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context);
static void clear_sip_domains(void);

/*--- SIP realm authentication */
static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno);
static int clear_realm_authentication(struct sip_auth *authlist);       /* Clear realm authentication list (at reload) */
static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm);

/*--- Misc functions */
static int sip_do_reload(enum channelreloadreason reason);
static int reload_config(enum channelreloadreason reason);
static int expire_register(void *data);
static void *do_monitor(void *data);
static int restart_monitor(void);
static int sip_send_mwi_to_peer(struct sip_peer *peer);
static void sip_destroy(struct sip_pvt *p);
static int sip_addrcmp(char *name, struct sockaddr_in *sin);    /* Support for peer matching */
static int sip_refer_allocate(struct sip_pvt *p);
static void ast_quiet_chan(struct ast_channel *chan);
static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);

/*--- Device monitoring and Device/extension state handling */
static int cb_extensionstate(char *context, char* exten, int state, void *data);
static int sip_devicestate(void *data);
static int sip_poke_noanswer(void *data);
static int sip_poke_peer(struct sip_peer *peer);
static void sip_poke_all_peers(void);
static void sip_peer_hold(struct sip_pvt *p, int hold);

/*--- Applications, functions, CLI and manager command helpers */
static const char *sip_nat_mode(const struct sip_pvt *p);
static int sip_show_inuse(int fd, int argc, char *argv[]);
static char *transfermode2str(enum transfermodes mode) attribute_const;
static char *nat2str(int nat) attribute_const;
static int peer_status(struct sip_peer *peer, char *status, int statuslen);
static int sip_show_users(int fd, int argc, char *argv[]);
static int _sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static int sip_show_peers(int fd, int argc, char *argv[]);
static int sip_show_objects(int fd, int argc, char *argv[]);
static void  print_group(int fd, ast_group_t group, int crlf);
static const char *dtmfmode2str(int mode) attribute_const;
static const char *insecure2str(int port, int invite) attribute_const;
static void cleanup_stale_contexts(char *new, char *old);
static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
static const char *domain_mode_to_text(const enum domain_mode mode);
static int sip_show_domains(int fd, int argc, char *argv[]);
static int _sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static int sip_show_peer(int fd, int argc, char *argv[]);
static int sip_show_user(int fd, int argc, char *argv[]);
static int sip_show_registry(int fd, int argc, char *argv[]);
static int sip_show_settings(int fd, int argc, char *argv[]);
static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions);
static int sip_show_channels(int fd, int argc, char *argv[]);
static int sip_show_subscriptions(int fd, int argc, char *argv[]);
static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions);
static char *complete_sipch(const char *line, const char *word, int pos, int state);
static char *complete_sip_peer(const char *word, int state, int flags2);
static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
static char *complete_sip_debug_peer(const char *line, const char *word, int pos, int state);
static char *complete_sip_user(const char *word, int state, int flags2);
static char *complete_sip_show_user(const char *line, const char *word, int pos, int state);
static char *complete_sipnotify(const char *line, const char *word, int pos, int state);
static char *complete_sip_prune_realtime_peer(const char *line, const char *word, int pos, int state);
static char *complete_sip_prune_realtime_user(const char *line, const char *word, int pos, int state);
static int sip_show_channel(int fd, int argc, char *argv[]);
static int sip_show_history(int fd, int argc, char *argv[]);
static int sip_do_debug_ip(int fd, int argc, char *argv[]);
static int sip_do_debug_peer(int fd, int argc, char *argv[]);
static int sip_do_debug(int fd, int argc, char *argv[]);
static int sip_no_debug(int fd, int argc, char *argv[]);
static int sip_notify(int fd, int argc, char *argv[]);
static int sip_do_history(int fd, int argc, char *argv[]);
static int sip_no_history(int fd, int argc, char *argv[]);
static int sip_dtmfmode(struct ast_channel *chan, void *data);
static int sip_addheader(struct ast_channel *chan, void *data);
static int sip_do_reload(enum channelreloadreason reason);
static int sip_reload(int fd, int argc, char *argv[]);

/*--- Debugging 
        Functions for enabling debug per IP or fully, or enabling history logging for
        a SIP dialog
*/
static void sip_dump_history(struct sip_pvt *dialog);   /* Dump history to LOG_DEBUG at end of dialog, before destroying data */
static inline int sip_debug_test_addr(const struct sockaddr_in *addr);
static inline int sip_debug_test_pvt(struct sip_pvt *p);
static void append_history_full(struct sip_pvt *p, const char *fmt, ...);
static void sip_dump_history(struct sip_pvt *dialog);

/*--- Device object handling */
static struct sip_peer *temp_peer(const char *name);
static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime);
static struct sip_user *build_user(const char *name, struct ast_variable *v, int realtime);
static int update_call_counter(struct sip_pvt *fup, int event);
static void sip_destroy_peer(struct sip_peer *peer);
static void sip_destroy_user(struct sip_user *user);
static int sip_poke_peer(struct sip_peer *peer);
static void set_peer_defaults(struct sip_peer *peer);
static struct sip_peer *temp_peer(const char *name);
static void register_peer_exten(struct sip_peer *peer, int onoff);
static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime);
static struct sip_user *find_user(const char *name, int realtime);
static int sip_poke_peer_s(void *data);
static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
static void reg_source_db(struct sip_peer *peer);
static void destroy_association(struct sip_peer *peer);
static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v);

/* Realtime device support */
static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey);
static struct sip_user *realtime_user(const char *username);
static void update_peer(struct sip_peer *p, int expiry);
static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin);
static int sip_prune_realtime(int fd, int argc, char *argv[]);

/*--- Internal UA client handling (outbound registrations) */
static int ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us);
static void sip_registry_destroy(struct sip_registry *reg);
static int sip_register(char *value, int lineno);
static char *regstate2str(enum sipregistrystate regstate) attribute_const;
static int sip_reregister(void *data);
static int __sip_do_register(struct sip_registry *r);
static int sip_reg_timeout(void *data);
static void sip_send_all_registers(void);

/*--- Parsing SIP requests and responses */
static void append_date(struct sip_request *req);       /* Append date to SIP packet */
static int determine_firstline_parts(struct sip_request *req);
static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
static int find_sip_method(const char *msg);
static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported);
static void parse_request(struct sip_request *req);
static const char *get_header(const struct sip_request *req, const char *name);
static const char *referstatus2str(enum referstatus rstatus) attribute_pure;
static int method_match(enum sipmethod id, const char *name);
static void parse_copy(struct sip_request *dst, const struct sip_request *src);
static char *get_in_brackets(char *tmp);
static const char *find_alias(const char *name, const char *_default);
static const char *__get_header(const struct sip_request *req, const char *name, int *start);
static int lws2sws(char *msgbuf, int len);
static void extract_uri(struct sip_pvt *p, struct sip_request *req);
static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req);
static int get_also_info(struct sip_pvt *p, struct sip_request *oreq);
static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req);
static int set_address_from_contact(struct sip_pvt *pvt);
static void check_via(struct sip_pvt *p, struct sip_request *req);
static char *get_calleridname(const char *input, char *output, size_t outputsize);
static int get_rpid_num(const char *input, char *output, int maxlen);
static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq);
static int get_destination(struct sip_pvt *p, struct sip_request *oreq);
static int get_msg_text(char *buf, int len, struct sip_request *req);
static void free_old_route(struct sip_route *route);
static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout);

/*--- Constructing requests and responses */
static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
static int init_req(struct sip_request *req, int sipmethod, const char *recip);
static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch);
static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod);
static int init_resp(struct sip_request *resp, const char *msg);
static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req);
static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p);
static void build_via(struct sip_pvt *p);
static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer);
static int create_addr(struct sip_pvt *dialog, const char *opeer);
static char *generate_random_string(char *buf, size_t size);
static void build_callid_pvt(struct sip_pvt *pvt);
static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain);
static void make_our_tag(char *tagbuf, size_t len);
static int add_header(struct sip_request *req, const char *var, const char *value);
static int add_header_contentLength(struct sip_request *req, int len);
static int add_line(struct sip_request *req, const char *line);
static int add_text(struct sip_request *req, const char *text);
static int add_digit(struct sip_request *req, char digit);
static int add_vidupdate(struct sip_request *req);
static void add_route(struct sip_request *req, struct sip_route *route);
static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field);
static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field);
static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field);
static void set_destination(struct sip_pvt *p, char *uri);
static void append_date(struct sip_request *req);
static void build_contact(struct sip_pvt *p);
static void build_rpid(struct sip_pvt *p);

/*------Request handling functions */
static int handle_request(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock);
static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *recount, char *e);
static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, int *nounlock);
static int handle_request_bye(struct sip_pvt *p, struct sip_request *req);
static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e);
static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
static int handle_request_message(struct sip_pvt *p, struct sip_request *req);
static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
static int handle_request_options(struct sip_pvt *p, struct sip_request *req);
static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin);
static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno);

/*------Response handling functions */
static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);

/*----- RTP interface functions */
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active);
static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
static int sip_get_codec(struct ast_channel *chan);
static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);

/*------ T38 Support --------- */
static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */
static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);

/*! \brief Definition of this channel for PBX channel registration */
static const struct ast_channel_tech sip_tech = {
        .type = "SIP",
        .description = "Session Initiation Protocol (SIP)",
        .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
        .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
        .requester = sip_request_call,
        .devicestate = sip_devicestate,
        .call = sip_call,
        .hangup = sip_hangup,
        .answer = sip_answer,
        .read = sip_read,
        .write = sip_write,
        .write_video = sip_write,
        .indicate = sip_indicate,
        .transfer = sip_transfer,
        .fixup = sip_fixup,
        .send_digit_begin = sip_senddigit_begin,
        .send_digit_end = sip_senddigit_end,
        .bridge = ast_rtp_bridge,
        .early_bridge = ast_rtp_early_bridge,
        .send_text = sip_sendtext,
};

/**--- some list management macros. **/
 
#define UNLINK(element, head, prev) do {        \
        if (prev)                               \
                (prev)->next = (element)->next; \
        else                                    \
                (head) = (element)->next;       \
        } while (0)

/*! \brief Interface structure with callbacks used to connect to RTP module */
static struct ast_rtp_protocol sip_rtp = {
        type: "SIP",
        get_rtp_info: sip_get_rtp_peer,
        get_vrtp_info: sip_get_vrtp_peer,
        set_rtp_peer: sip_set_rtp_peer,
        get_codec: sip_get_codec,
};

/*! \brief Helper function to lock, hiding the underlying locking mechanism.  */
static void sip_pvt_lock(struct sip_pvt *pvt)
{
        ast_mutex_lock(&pvt->pvt_lock);
}

/*! \brief Helper function to unlock pvt, hiding the underlying locking mechanism. */
static void sip_pvt_unlock(struct sip_pvt *pvt)
{
        ast_mutex_unlock(&pvt->pvt_lock);
}

/*!
 * helper functions to unreference various types of objects.
 * By handling them this way, we don't have to declare the
 * destructor on each call, which removes the chance of errors.
 */
static void unref_peer(struct sip_peer *peer)
{
        ASTOBJ_UNREF(peer, sip_destroy_peer);
}

static void unref_user(struct sip_user *user)
{
        ASTOBJ_UNREF(user, sip_destroy_user);
}

static void unref_registry(struct sip_registry *reg)
{
        ASTOBJ_UNREF(reg, sip_registry_destroy);
}

/*! \brief Interface structure with callbacks used to connect to UDPTL module*/
static struct ast_udptl_protocol sip_udptl = {
        type: "SIP",
        get_udptl_info: sip_get_udptl_peer,
        set_udptl_peer: sip_set_udptl_peer,
};

/*! \brief Convert transfer status to string */
static const char *referstatus2str(enum referstatus rstatus)
{
        int i = (sizeof(referstatusstrings) / sizeof(referstatusstrings[0]));
        int x;

        for (x = 0; x < i; x++) {
                if (referstatusstrings[x].status ==  rstatus)
                        return referstatusstrings[x].text;
        }
        return "";
}

/*! \brief Initialize the initital request packet in the pvt structure.
        This packet is used for creating replies and future requests in
        a dialog */
static void initialize_initreq(struct sip_pvt *p, struct sip_request *req)
{
        if (option_debug) {
                if (p->initreq.headers)
                        ast_log(LOG_DEBUG, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
                else
                        ast_log(LOG_DEBUG, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
        }
        /* Use this as the basis */
        copy_request(&p->initreq, req);
        parse_request(&p->initreq);
        if (ast_test_flag(req, SIP_PKT_DEBUG))
                ast_verbose("Initreq: %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}

/*! \brief Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging */
static void sip_alreadygone(struct sip_pvt *dialog)
{
        if (option_debug > 2)
                ast_log(LOG_DEBUG, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
        ast_set_flag(&dialog->flags[0], SIP_ALREADYGONE);
}


/*! \brief returns true if 'name' (with optional trailing whitespace)
 * matches the sip method 'id'.
 * Strictly speaking, SIP methods are case SENSITIVE, but we do
 * a case-insensitive comparison to be more tolerant.
 * following Jon Postel's rule: Be gentle in what you accept, strict with what you send
 */
static int method_match(enum sipmethod id, const char *name)
{
        int len = strlen(sip_methods[id].text);
        int l_name = name ? strlen(name) : 0;
        /* true if the string is long enough, and ends with whitespace, and matches */
        return (l_name >= len && name[len] < 33 &&
                !strncasecmp(sip_methods[id].text, name, len));
}

/*! \brief  find_sip_method: Find SIP method from header */
static int find_sip_method(const char *msg)
{
        int i, res = 0;
        
        if (ast_strlen_zero(msg))
                return 0;
        for (i = 1; i < (sizeof(sip_methods) / sizeof(sip_methods[0])) && !res; i++) {
                if (method_match(i, msg))
                        res = sip_methods[i].id;
        }
        return res;
}

/*! \brief Parse supported header in incoming packet */
static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported)
{
        char *next, *sep;
        char *temp;
        unsigned int profile = 0;
        int i, found;

        if (ast_strlen_zero(supported) )
                return 0;
        temp = ast_strdupa(supported);

        if (option_debug > 2 && sipdebug)
                ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported);

        for (next = temp; next; next = sep) {
                found = FALSE;
                if ( (sep = strchr(next, ',')) != NULL)
                        *sep++ = '\0';
                next = ast_skip_blanks(next);
                if (option_debug > 2 && sipdebug)
                        ast_log(LOG_DEBUG, "Found SIP option: -%s-\n", next);
                for (i=0; i < (sizeof(sip_options) / sizeof(sip_options[0])); i++) {
                        if (!strcasecmp(next, sip_options[i].text)) {
                                profile |= sip_options[i].id;
                                found = TRUE;
                                if (option_debug > 2 && sipdebug)
                                        ast_log(LOG_DEBUG, "Matched SIP option: %s\n", next);
                                break;
                        }
                }
                if (!found && option_debug > 2 && sipdebug) {
                        if (!strncasecmp(next, "x-", 2))
                                ast_log(LOG_DEBUG, "Found private SIP option, not supported: %s\n", next);
                        else
                                ast_log(LOG_DEBUG, "Found no match for SIP option: %s (Please file bug report!)\n", next);
                }
        }

        if (pvt)
                pvt->sipoptions = profile;
        return profile;
}

/*! \brief See if we pass debug IP filter */
static inline int sip_debug_test_addr(const struct sockaddr_in *addr) 
{
        if (!sipdebug)
                return 0;
        if (debugaddr.sin_addr.s_addr) {
                if (((ntohs(debugaddr.sin_port) != 0)
                        && (debugaddr.sin_port != addr->sin_port))
                        || (debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
                        return 0;
        }
        return 1;
}

/*! \brief The real destination address for a write */
static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p)
{
        return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? &p->recv : &p->sa;
}

/*! \brief Display SIP nat mode */
static const char *sip_nat_mode(const struct sip_pvt *p)
{
        return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? "NAT" : "no NAT";
}

/*! \brief Test PVT for debugging output */
static inline int sip_debug_test_pvt(struct sip_pvt *p) 
{
        if (!sipdebug)
                return 0;
        return sip_debug_test_addr(sip_real_dst(p));
}

/*! \brief Transmit SIP message */
static int __sip_xmit(struct sip_pvt *p, char *data, int len)
{
        int res;
        const struct sockaddr_in *dst = sip_real_dst(p);
        res = sendto(sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));

        if (res != len)
                ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno));
        return res;
}


/*! \brief Build a Via header for a request */
static void build_via(struct sip_pvt *p)
{
        /* Work around buggy UNIDEN UIP200 firmware */
        const char *rport = ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_RFC3581 ? ";rport" : "";

        /* z9hG4bK is a magic cookie.  See RFC 3261 section 8.1.1.7 */
        ast_string_field_build(p, via, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x%s",
                         ast_inet_ntoa(p->ourip), ourport, p->branch, rport);
}

/*! \brief NAT fix - decide which IP address to use for ASterisk server?
 *
 * Using the localaddr structure built up with localnet statements in sip.conf
 * apply it to their address to see if we need to substitute our
 * externip or can get away with our internal bindaddr
 */
static enum sip_result ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us)
{
        struct sockaddr_in theirs, ours;

        /* Get our local information */
        ast_ouraddrfor(them, us);
        theirs.sin_addr = *them;
        ours.sin_addr = *us;

        if (localaddr && externip.sin_addr.s_addr &&
            ast_apply_ha(localaddr, &theirs) &&
            !ast_apply_ha(localaddr, &ours)) {
                if (externexpire && time(NULL) >= externexpire) {
                        struct ast_hostent ahp;
                        struct hostent *hp;

                        externexpire = time(NULL) + externrefresh;
                        if ((hp = ast_gethostbyname(externhost, &ahp))) {
                                memcpy(&externip.sin_addr, hp->h_addr, sizeof(externip.sin_addr));
                        } else
                                ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
                }
                *us = externip.sin_addr;
                if (option_debug) {
                        ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n", 
                                ast_inet_ntoa(*(struct in_addr *)&them->s_addr));
                }
        } else if (bindaddr.sin_addr.s_addr)
                *us = bindaddr.sin_addr;
        return AST_SUCCESS;
}

/*! \brief Append to SIP dialog history 
        \return Always returns 0 */
#define append_history(p, event, fmt , args... )        append_history_full(p, "%-15s " fmt, event, ## args)

static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
        __attribute__ ((format (printf, 2, 3)));

/*! \brief Append to SIP dialog history with arg list  */
static void append_history_va(struct sip_pvt *p, const char *fmt, va_list ap)
{
        char buf[80], *c = buf; /* max history length */
        struct sip_history *hist;
        int l;

        vsnprintf(buf, sizeof(buf), fmt, ap);
        strsep(&c, "\r\n"); /* Trim up everything after \r or \n */
        l = strlen(buf) + 1;
        if (!(hist = ast_calloc(1, sizeof(*hist) + l)))
                return;
        if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
                free(hist);
                return;
        }
        memcpy(hist->event, buf, l);
        AST_LIST_INSERT_TAIL(p->history, hist, list);
}

/*! \brief Append to SIP dialog history with arg list  */
static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
{
        va_list ap;

        if (!p)
                return;
        va_start(ap, fmt);
        append_history_va(p, fmt, ap);
        va_end(ap);

        return;
}

/*! \brief Retransmit SIP message if no answer (Called from scheduler) */
static int retrans_pkt(void *data)
{
        struct sip_pkt *pkt = data, *prev, *cur = NULL;
        int reschedule = DEFAULT_RETRANS;

        /* Lock channel PVT */
        sip_pvt_lock(pkt->owner);

        if (pkt->retrans < MAX_RETRANS) {
                pkt->retrans++;
                if (!pkt->timer_t1) {   /* Re-schedule using timer_a and timer_t1 */
                        if (sipdebug && option_debug > 3)
                                ast_log(LOG_DEBUG, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n", pkt->retransid, sip_methods[pkt->method].text, pkt->method);
                } else {
                        int siptimer_a;

                        if (sipdebug && option_debug > 3)
                                ast_log(LOG_DEBUG, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method);
                        if (!pkt->timer_a)
                                pkt->timer_a = 2 ;
                        else
                                pkt->timer_a = 2 * pkt->timer_a;
 
                        /* For non-invites, a maximum of 4 secs */
                        siptimer_a = pkt->timer_t1 * pkt->timer_a;      /* Double each time */
                        if (pkt->method != SIP_INVITE && siptimer_a > 4000)
                                siptimer_a = 4000;
                
                        /* Reschedule re-transmit */
                        reschedule = siptimer_a;
                        if (option_debug > 3)
                                ast_log(LOG_DEBUG, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n", pkt->retrans +1, siptimer_a, pkt->timer_t1, pkt->retransid);
                } 

                if (sip_debug_test_pvt(pkt->owner)) {
                        const struct sockaddr_in *dst = sip_real_dst(pkt->owner);
                        ast_verbose("Retransmitting #%d (%s) to %s:%d:\n%s\n---\n",
                                pkt->retrans, sip_nat_mode(pkt->owner),
                                ast_inet_ntoa(dst->sin_addr),
                                ntohs(dst->sin_port), pkt->data);
                }

                append_history(pkt->owner, "ReTx", "%d %s", reschedule, pkt->data);
                __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
                sip_pvt_unlock(pkt->owner);
                return  reschedule;
        } 
        /* Too many retries */
        if (pkt->owner && pkt->method != SIP_OPTIONS) {
                if (ast_test_flag(pkt, FLAG_FATAL) || sipdebug) /* Tell us if it's critical or if we're debugging */
                        ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (ast_test_flag(pkt, FLAG_FATAL)) ? "Critical" : "Non-critical", (ast_test_flag(pkt, FLAG_RESPONSE)) ? "Response" : "Request");
        } else {
                if ((pkt->method == SIP_OPTIONS) && sipdebug)
                        ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) \n", pkt->owner->callid);
        }
        append_history(pkt->owner, "MaxRetries", "%s", (ast_test_flag(pkt, FLAG_FATAL)) ? "(Critical)" : "(Non-critical)");
                
        pkt->retransid = -1;

        if (ast_test_flag(pkt, FLAG_FATAL)) {
                while(pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) {
                        sip_pvt_unlock(pkt->owner);     /* SIP_PVT, not channel */
                        usleep(1);
                        sip_pvt_lock(pkt->owner);
                }
                if (pkt->owner->owner) {
                        sip_alreadygone(pkt->owner);
                        ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid);
                        ast_queue_hangup(pkt->owner->owner);
                        ast_channel_unlock(pkt->owner->owner);
                } else {
                        /* If no channel owner, destroy now */

                        /* Let the peerpoke system expire packets when the timer expires for poke_noanswer */
                        if (pkt->method != SIP_OPTIONS)
                                ast_set_flag(&pkt->owner->flags[0], SIP_NEEDDESTROY);   
                }
        }
        /* Remove the packet */
        for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) {
                if (cur == pkt) {
                        UNLINK(cur, pkt->owner->packets, prev);
                        sip_pvt_unlock(pkt->owner);
                        free(pkt);
                        return 0;
                }
        }
        /* error case */
        ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
        sip_pvt_unlock(pkt->owner);
        return 0;
}

/*! \brief Transmit packet with retransmits 
        \return 0 on success, -1 on failure to allocate packet 
*/
static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod)
{
        struct sip_pkt *pkt;
        int siptimer_a = DEFAULT_RETRANS;

        if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1)))
                return AST_FAILURE;
        memcpy(pkt->data, data, len);
        pkt->method = sipmethod;
        pkt->packetlen = len;
        pkt->next = p->packets;
        pkt->owner = p;
        pkt->seqno = seqno;
        if (resp)
                ast_set_flag(pkt, FLAG_RESPONSE);
        pkt->data[len] = '\0';
        pkt->timer_t1 = p->timer_t1;    /* Set SIP timer T1 */
        if (fatal)
                ast_set_flag(pkt, FLAG_FATAL);
        if (pkt->timer_t1)
                siptimer_a = pkt->timer_t1 * 2;

        /* Schedule retransmission */
        pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1);
        if (option_debug > 3 && sipdebug)
                ast_log(LOG_DEBUG, "*** SIP TIMER: Initalizing retransmit timer on packet: Id  #%d\n", pkt->retransid);
        pkt->next = p->packets;
        p->packets = pkt;

        __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);      /* Send packet */
        if (sipmethod == SIP_INVITE) {
                /* Note this is a pending invite */
                p->pendinginvite = seqno;
        }
        return AST_SUCCESS;
}

/*! \brief Kill a SIP dialog (called by scheduler) */
static int __sip_autodestruct(void *data)
{
        struct sip_pvt *p = data;

        /* If this is a subscription, tell the phone that we got a timeout */
        if (p->subscribed) {
                transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1, TRUE);   /* Send last notification */
                p->subscribed = NONE;
                append_history(p, "Subscribestatus", "timeout");
                if (option_debug > 2)
                        ast_log(LOG_DEBUG, "Re-scheduled destruction of SIP subsription %s\n", p->callid ? p->callid : "<unknown>");
                return 10000;   /* Reschedule this destruction so that we know that it's gone */
        }

        if (p->subscribed == MWI_NOTIFICATION)
                if (p->relatedpeer)
                        unref_peer(p->relatedpeer);     /* Remove link to peer. If it's realtime, make sure it's gone from memory) */

        /* Reset schedule ID */
        p->autokillid = -1;

        if (p->owner) {
                ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
                ast_queue_hangup(p->owner);
        } else if (p->refer) {
                if (option_debug > 2)
                        ast_log(LOG_DEBUG, "Finally hanging up channel after transfer: %s\n", p->callid);
                transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
                append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
        } else {
                append_history(p, "AutoDestroy", "%s", p->callid);
                if (option_debug)
                        ast_log(LOG_DEBUG, "Auto destroying SIP dialog '%s'\n", p->callid);
                sip_destroy(p);         /* Go ahead and destroy dialog. All attempts to recover is done */
        }
        return 0;
}

/*! \brief Schedule destruction of SIP dialog */
static void sip_scheddestroy(struct sip_pvt *p, int ms)
{
        if (ms < 0) {
                if (p->timer_t1 == 0)
                        p->timer_t1 = SIP_TIMER_T1;     /* Set timer T1 if not set (RFC 3261) */
                ms = p->timer_t1 * 64;
        }
        if (sip_debug_test_pvt(p))
                ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
        if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
                append_history(p, "SchedDestroy", "%d ms", ms);

        if (p->autokillid > -1)
                ast_sched_del(sched, p->autokillid);
        p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, p);
}

/*! \brief Cancel destruction of SIP dialog */
static void sip_cancel_destroy(struct sip_pvt *p)
{
        if (p->autokillid > -1) {
                ast_sched_del(sched, p->autokillid);
                append_history(p, "CancelDestroy", "");
                p->autokillid = -1;
        }
}

/*! \brief Acknowledges receipt of a packet and stops retransmission */
static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
{
        struct sip_pkt *cur, *prev = NULL;
        const char *msg = "Not Found";  /* used only for debugging */

        sip_pvt_lock(p);
        for (cur = p->packets; cur; prev = cur, cur = cur->next) {
                if (cur->seqno != seqno || ast_test_flag(cur, FLAG_RESPONSE) != resp)
                        continue;
                if (ast_test_flag(cur, FLAG_RESPONSE) || cur->method == sipmethod) {
                        msg = "Found";
                        if (!resp && (seqno == p->pendinginvite)) {
                                if (option_debug)
                                        ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite);
                                p->pendinginvite = 0;
                        }
                        if (cur->retransid > -1) {
                                if (sipdebug && option_debug > 3)
                                        ast_log(LOG_DEBUG, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
                                ast_sched_del(sched, cur->retransid);
                                cur->retransid = -1;
                        }
                        UNLINK(cur, p->packets, prev);
                        free(cur);
                        break;
                }
        }
        sip_pvt_unlock(p);
        if (option_debug)
                ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %s\n",
                        p->callid, resp ? "Response" : "Request", seqno, msg);
}

/*! \brief Pretend to ack all packets
 * maybe the lock on p is not strictly necessary but there might be a race */
static void __sip_pretend_ack(struct sip_pvt *p)
{
        struct sip_pkt *cur = NULL;

        while (p->packets) {
                int method;
                if (cur == p->packets) {
                        ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
                        return;
                }
                cur = p->packets;
                method = (cur->method) ? cur->method : find_sip_method(cur->data);
                __sip_ack(p, cur->seqno, ast_test_flag(cur, FLAG_RESPONSE), method);
        }
}

/*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
{
        struct sip_pkt *cur;
        int res = -1;

        for (cur = p->packets; cur; cur = cur->next) {
                if (cur->seqno == seqno && ast_test_flag(cur, FLAG_RESPONSE) == resp &&
                        (ast_test_flag(cur, FLAG_RESPONSE) || method_match(sipmethod, cur->data))) {
                        /* this is our baby */
                        if (cur->retransid > -1) {
                                if (option_debug > 3 && sipdebug)
                                        ast_log(LOG_DEBUG, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
                                ast_sched_del(sched, cur->retransid);
                                cur->retransid = -1;
                        }
                        res = 0;
                        break;
                }
        }
        if (option_debug)
                ast_log(LOG_DEBUG, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
        return res;
}


/*! \brief Copy SIP request, parse it */
static void parse_copy(struct sip_request *dst, const struct sip_request *src)
{
        memset(dst, 0, sizeof(*dst));
        memcpy(dst->data, src->data, sizeof(dst->data));
        dst->len = src->len;
        parse_request(dst);
}

/*! \brief add a blank line if no body */
static void add_blank(struct sip_request *req)
{
        if (!req->lines) {
                /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
                snprintf(req->data + req->len, sizeof(req->data) - req->len, "\r\n");
                req->len += strlen(req->data + req->len);
        }
}

/*! \brief Transmit response on SIP request*/
static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
{
        int res;

        add_blank(req);
        if (sip_debug_test_pvt(p)) {
                const struct sockaddr_in *dst = sip_real_dst(p);

                ast_verbose("\n<--- %sTransmitting (%s) to %s:%d --->\n%s\n<------------>\n",
                        reliable ? "Reliably " : "", sip_nat_mode(p),
                        ast_inet_ntoa(dst->sin_addr),
                        ntohs(dst->sin_port), req->data);
        }
        if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
                struct sip_request tmp;
                parse_copy(&tmp, req);
                append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), 
                        (tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? tmp.rlPart2 : sip_methods[tmp.method].text);
        }
        res = (reliable) ?
                __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
                __sip_xmit(p, req->data, req->len);
        if (res > 0)
                return 0;
        return res;
}

/*! \brief Send SIP Request to the other part of the dialogue */
static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
{
        int res;

        add_blank(req);
        if (sip_debug_test_pvt(p)) {
                if (ast_test_flag(&p->flags[0], SIP_NAT_ROUTE))
                        ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port), req->data);
                else
                        ast_verbose("%sTransmitting (no NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->sa.sin_addr), ntohs(p->sa.sin_port), req->data);
        }
        if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
                struct sip_request tmp;
                parse_copy(&tmp, req);
                append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
        }
        res = (reliable) ?
                __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1), req->method) :
                __sip_xmit(p, req->data, req->len);
        return res;
}

/*! \brief Locate closing quote in a string, skipping escaped quotes.
 * optionally with a limit on the search.
 * start must be past the first quote.
 */
static const char *find_closing_quote(const char *start, const char *lim)
{
        char last_char = '\0';
        const char *s;
        for (s = start; *s && s != lim; last_char = *s++) {
                if (*s == '"' && last_char != '\\')
                        break;
        }
        return s;
}

/*! \brief Pick out text in brackets from character string
        \return pointer to terminated stripped string
        \param tmp input string that will be modified
        Examples:

        "foo" <bar>     valid input, returns bar
        foo             returns the whole string
        < "foo ... >    returns the string between brackets
        < "foo...       bogus (missing closing bracket), returns the whole string
                        XXX maybe should still skip the opening bracket
 */
static char *get_in_brackets(char *tmp)
{
        const char *parse = tmp;
        char *first_bracket;

        /*
         * Skip any quoted text until we find the part in brackets.
         * On any error give up and return the full string.
         */
        while ( (first_bracket = strchr(parse, '<')) ) {
                char *first_quote = strchr(parse, '"');

                if (!first_quote || first_quote > first_bracket)
                        break; /* no need to look at quoted part */
                /* the bracket is within quotes, so ignore it */
                parse = find_closing_quote(first_quote + 1, NULL);
                if (!*parse) { /* not found, return full string ? */
                        /* XXX or be robust and return in-bracket part ? */
                        ast_log(LOG_WARNING, "No closing quote found in '%s'\n", tmp);
                        break;
                }
                parse++;
        }
        if (first_bracket) {
                char *second_bracket = strchr(first_bracket + 1, '>');
                if (second_bracket) {
                        *second_bracket = '\0';
                        tmp = first_bracket + 1;
                } else {
                        ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp);
                }
        }
        return tmp;
}

/*!
 * parses a URI in its components.
 * If scheme is specified, drop it from the top.
 * If a component is not requested, do not split around it.
 * This means that if we don't have domain, we cannot split
 * name:pass and domain:port.
 * It is safe to call with ret_name, pass, domain, port
 * pointing all to the same place.
 * Init pointers to empty string so we never get NULL dereferencing.
 * Overwrites the string.
 * return 0 on success, other values on error.
 */
static int parse_uri(char *uri, char *scheme,
        char **ret_name, char **pass, char **domain, char **port, char **options)
{
        char *name = NULL;
        int error = 0;

        /* init field as required */
        if (pass)
                *pass = "";
        if (port)
                *port = "";
        name = strsep(&uri, ";");       /* remove options */
        if (scheme) {
                int l = strlen(scheme);
                if (!strncmp(name, scheme, l))
                        name += l;
                else {
                        ast_log(LOG_NOTICE, "Missing scheme '%s' in '%s'\n", scheme, name);
                        error = -1;
                }
        }
        if (!domain) {
                /* if we don't want to split around domain, keep everything as a name,
                 * so we need to do nothing here, except remember why.
                 */
        } else {
                /* store the result in a temp. variable to avoid it being
                 * overwritten if arguments point to the same place.
                 */
                char *c, *dom = "";

                if ((c = strchr(name, '@')) == NULL) {
                        /* domain-only URI, according to the SIP RFC. */
                        dom = name;
                        name = "";
                } else {
                        *c++ = '\0';
                        dom = c;
                }
                if (port && (c = strchr(dom, ':'))) { /* Remove :port */
                        *c++ = '\0';
                        *port = c;
                }
                if (pass && (c = strchr(name, ':'))) {  /* user:password */
                        *c++ = '\0';
                        *pass = c;
                }
                *domain = dom;
        }
        if (ret_name)   /* same as for domain, store the result only at the end */
                *ret_name = name;
        if (options)
                *options = uri ? uri : "";

        return error;
}

/*! \brief Send SIP MESSAGE text within a call
        Called from PBX core sendtext() application */
static int sip_sendtext(struct ast_channel *ast, const char *text)
{
        struct sip_pvt *p = ast->tech_pvt;
        int debug = sip_debug_test_pvt(p);

        if (debug)
                ast_verbose("Sending text %s on %s\n", text, ast->name);
        if (!p)
                return -1;
        if (ast_strlen_zero(text))
                return 0;
        if (debug)
                ast_verbose("Really sending text %s on %s\n", text, ast->name);
        transmit_message_with_text(p, text);
        return 0;       
}

/*! \brief Update peer object in realtime storage 
        If the Asterisk system name is set in asterisk.conf, we will use
        that name and store that in the "regserver" field in the sippeers
        table to facilitate multi-server setups.
*/
static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey)
{
        char port[10];
        char ipaddr[INET_ADDRSTRLEN];
        char regseconds[20];

        char *sysname = ast_config_AST_SYSTEM_NAME;
        char *syslabel = NULL;

        time_t nowtime = time(NULL) + expirey;
        const char *fc = fullcontact ? "fullcontact" : NULL;
        
        snprintf(regseconds, sizeof(regseconds), "%d", (int)nowtime);   /* Expiration time */
        ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
        snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port));
        
        if (ast_strlen_zero(sysname))   /* No system name, disable this */
                sysname = NULL;
        else if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTSAVE_SYSNAME))
                syslabel = "regserver";

        if (fc)
                ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr,
                        "port", port, "regseconds", regseconds,
                        "username", username, fc, fullcontact, syslabel, sysname, NULL); /* note fc and syslabel _can_ be NULL */
        else
                ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr,
                        "port", port, "regseconds", regseconds,
                        "username", username, syslabel, sysname, NULL); /* note syslabel _can_ be NULL */
}

/*! \brief Automatically add peer extension to dial plan */
static void register_peer_exten(struct sip_peer *peer, int onoff)
{
        char multi[256];
        char *stringp, *ext, *context;

        /* XXX note that global_regcontext is both a global 'enable' flag and
         * the name of the global regexten context, if not specified
         * individually.
         */
        if (ast_strlen_zero(global_regcontext))
                return;

        ast_copy_string(multi, S_OR(peer->regexten, peer->name), sizeof(multi));
        stringp = multi;
        while ((ext = strsep(&stringp, "&"))) {
                if ((context = strchr(ext, '@'))) {
                        *context++ = '\0';      /* split ext@context */
                        if (!ast_context_find(context)) {
                                ast_log(LOG_WARNING, "Context %s must exist in regcontext= in sip.conf!\n", context);
                                continue;
                        }
                } else {
                        context = global_regcontext;
                }
                if (onoff)
                        ast_add_extension(context, 1, ext, 1, NULL, NULL, "Noop",
                                 ast_strdup(peer->name), ast_free, "SIP");
                else
                        ast_context_remove_extension(context, ext, 1, NULL);
        }
}

/*! \brief Destroy peer object from memory */
static void sip_destroy_peer(struct sip_peer *peer)
{
        if (option_debug > 2)
                ast_log(LOG_DEBUG, "Destroying SIP peer %s\n", peer->name);

        /* Delete it, it needs to disappear */
        if (peer->call)
                sip_destroy(peer->call);

        if (peer->mwipvt)       /* We have an active subscription, delete it */
                sip_destroy(peer->mwipvt);

        if (peer->chanvars) {
                ast_variables_destroy(peer->chanvars);
                peer->chanvars = NULL;
        }
        if (peer->expire > -1)
                ast_sched_del(sched, peer->expire);

        if (peer->pokeexpire > -1)
                ast_sched_del(sched, peer->pokeexpire);
        register_peer_exten(peer, FALSE);
        ast_free_ha(peer->ha);
        if (ast_test_flag(&peer->flags[1], SIP_PAGE2_SELFDESTRUCT))
                apeerobjs--;
        else if (ast_test_flag(&peer->flags[0], SIP_REALTIME)) {
                rpeerobjs--;
                if (option_debug > 2)
                        ast_log(LOG_DEBUG,"-REALTIME- peer Destroyed. Name: %s. Realtime Peer objects: %d\n", peer->name, rpeerobjs);
        } else
                speerobjs--;
        clear_realm_authentication(peer->auth);
        peer->auth = NULL;
        if (peer->dnsmgr)
                ast_dnsmgr_release(peer->dnsmgr);
        free(peer);
}

/*! \brief Update peer data in database (if used) */
static void update_peer(struct sip_peer *p, int expiry)
{
        int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
        if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTUPDATE) &&
            (ast_test_flag(&p->flags[0], SIP_REALTIME) || rtcachefriends)) {
                realtime_update_peer(p->name, &p->addr, p->username, rtcachefriends ? p->fullcontact : NULL, expiry);
        }
}


/*! \brief  realtime_peer: Get peer from realtime storage
 * Checks the "sippeers" realtime family from extconfig.conf 
 * \todo Consider adding check of port address when matching here to follow the same
 *      algorithm as for static peers. Will we break anything by adding that?
*/
static struct sip_peer *realtime_peer(const char *newpeername, struct sockaddr_in *sin)
{
        struct sip_peer *peer;
        struct ast_variable *var = NULL;
        struct ast_variable *tmp;
        char ipaddr[INET_ADDRSTRLEN];

        /* First check on peer name */
        if (newpeername) 
                var = ast_load_realtime("sippeers", "name", newpeername, NULL);
        else if (sin) { /* Then check on IP address for dynamic peers */
                ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
                var = ast_load_realtime("sippeers", "host", ipaddr, NULL);      /* First check for fixed IP hosts */
                if (!var)
                        var = ast_load_realtime("sippeers", "ipaddr", ipaddr, NULL);    /* Then check for registred hosts */
        }

        if (!var)
                return NULL;

        for (tmp = var; tmp; tmp = tmp->next) {
                /* If this is type=user, then skip this object. */
                if (!strcasecmp(tmp->name, "type") &&
                    !strcasecmp(tmp->value, "user")) {
                        ast_variables_destroy(var);
                        return NULL;
                } else if (!newpeername && !strcasecmp(tmp->name, "name")) {
                        newpeername = tmp->value;
                }
        }
        
        if (!newpeername) {     /* Did not find peer in realtime */
                ast_log(LOG_WARNING, "Cannot Determine peer name ip=%s\n", ipaddr);
                ast_variables_destroy(var);
                return NULL;
        }


        /* Peer found in realtime, now build it in memory */
        peer = build_peer(newpeername, var, NULL, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
        if (!peer) {
                ast_variables_destroy(var);
                return NULL;
        }

        if (option_debug > 2)
                ast_log(LOG_DEBUG,"-REALTIME- loading peer from database to memory. Name: %s. Peer objects: %d\n", peer->name, rpeerobjs);

        if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
                /* Cache peer */
                ast_copy_flags(&peer->flags[1],&global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
                if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
                        if (peer->expire > -1) {
                                ast_sched_del(sched, peer->expire);
                        }
                        peer->expire = ast_sched_add(sched, (global_rtautoclear) * 1000, expire_register, (void *)peer);
                }
                ASTOBJ_CONTAINER_LINK(&peerl,peer);
        } else {
                ast_set_flag(&peer->flags[0], SIP_REALTIME);
        }
        ast_variables_destroy(var);

        return peer;
}

/*! \brief Support routine for find_peer */
static int sip_addrcmp(char *name, struct sockaddr_in *sin)
{
        /* We know name is the first field, so we can cast */
        struct sip_peer *p = (struct sip_peer *) name;
        return  !(!inaddrcmp(&p->addr, sin) || 
                                        (ast_test_flag(&p->flags[0], SIP_INSECURE_PORT) &&
                                        (p->addr.sin_addr.s_addr == sin->sin_addr.s_addr)));
}

/*! \brief Locate peer by name or ip address 
 *      This is used on incoming SIP message to find matching peer on ip
        or outgoing message to find matching peer on name */
static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime)
{
        struct sip_peer *p = NULL;

        if (peer)
                p = ASTOBJ_CONTAINER_FIND(&peerl, peer);
        else
                p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp);

        if (!p && realtime)
                p = realtime_peer(peer, sin);

        return p;
}

/*! \brief Remove user object from in-memory storage */
static void sip_destroy_user(struct sip_user *user)
{
        if (option_debug > 2)
                ast_log(LOG_DEBUG, "Destroying user object from memory: %s\n", user->name);
        ast_free_ha(user->ha);
        if (user->chanvars) {
                ast_variables_destroy(user->chanvars);
                user->chanvars = NULL;
        }
        if (ast_test_flag(&user->flags[0], SIP_REALTIME))
                ruserobjs--;
        else
                suserobjs--;
        free(user);
}

/*! \brief Load user from realtime storage
 * Loads user from "sipusers" category in realtime (extconfig.conf)
 * Users are matched on From: user name (the domain in skipped) */
static struct sip_user *realtime_user(const char *username)
{
        struct ast_variable *var;
        struct ast_variable *tmp;
        struct sip_user *user = NULL;

        var = ast_load_realtime("sipusers", "name", username, NULL);

        if (!var)
                return NULL;

        for (tmp = var; tmp; tmp = tmp->next) {
                if (!strcasecmp(tmp->name, "type") &&
                        !strcasecmp(tmp->value, "peer")) {
                        ast_variables_destroy(var);
                        return NULL;
                }
        }

        user = build_user(username, var, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
        
        if (!user) {    /* No user found */
                ast_variables_destroy(var);
                return NULL;
        }

        if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
                ast_set_flag(&user->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
                suserobjs++;
                ASTOBJ_CONTAINER_LINK(&userl,user);
        } else {
                /* Move counter from s to r... */
                suserobjs--;
                ruserobjs++;
                ast_set_flag(&user->flags[0], SIP_REALTIME);
        }
        ast_variables_destroy(var);
        return user;
}

/*! \brief Locate user by name 
 * Locates user by name (From: sip uri user name part) first
 * from in-memory list (static configuration) then from 
 * realtime storage (defined in extconfig.conf) */
static struct sip_user *find_user(const char *name, int realtime)
{
        struct sip_user *u = ASTOBJ_CONTAINER_FIND(&userl, name);
        if (!u && realtime)
                u = realtime_user(name);
        return u;
}

/*! \brief Set nat mode on the various data sockets */
static void do_setnat(struct sip_pvt *p, int natflags)
{
        const char *mode = natflags ? "On" : "Off";

        if (p->rtp) {
                if (option_debug)
                        ast_log(LOG_DEBUG, "Setting NAT on RTP to %s\n", mode);
                ast_rtp_setnat(p->rtp, natflags);
        }
        if (p->vrtp) {
                if (option_debug)
                        ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", mode);
                ast_rtp_setnat(p->vrtp, natflags);
        }
        if (p->udptl) {
                if (option_debug)
                        ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", mode);
                ast_udptl_setnat(p->udptl, natflags);
        }
}

/*! \brief Create address structure from peer reference.
 *  return -1 on error, 0 on success.
 */
static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
{
        if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) &&
            (!peer->maxms || ((peer->lastms >= 0)  && (peer->lastms <= peer->maxms)))) {
                dialog->sa = (peer->addr.sin_addr.s_addr) ? peer->addr : peer->defaddr;
                dialog->recv = dialog->sa;
        } else 
                return -1;

        ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
        ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
        dialog->capability = peer->capability;
        if ((!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) || !(dialog->capability & AST_FORMAT_VIDEO_MASK)) && dialog->vrtp) {
                ast_rtp_destroy(dialog->vrtp);
                dialog->vrtp = NULL;
        }
        dialog->prefs = peer->prefs;
        if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
                dialog->t38.capability = global_t38_capability;
                if (dialog->udptl) {
                        if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_FEC )
                                dialog->t38.capability |= T38FAX_UDP_EC_FEC;
                        else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY )
                                dialog->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
                        else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_NONE )
                                dialog->t38.capability |= T38FAX_UDP_EC_NONE;
                        dialog->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
                        if (option_debug > 1)
                                ast_log(LOG_DEBUG,"Our T38 capability (%d)\n", dialog->t38.capability);
                }
                dialog->t38.jointcapability = dialog->t38.capability;
        } else if (dialog->udptl) {
                ast_udptl_destroy(dialog->udptl);
                dialog->udptl = NULL;
        }
        do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE );

        if (dialog->rtp) {
                ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
                ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
                ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
                ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
                ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive);
                /* Set Frame packetization */
                ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
                dialog->autoframing = peer->autoframing;
        }
        if (dialog->vrtp) {
                ast_rtp_setdtmf(dialog->vrtp, 0);
                ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
                ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
                ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
                ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
        }

        ast_string_field_set(dialog, peername, peer->username);
        ast_string_field_set(dialog, authname, peer->username);
        ast_string_field_set(dialog, username, peer->username);
        ast_string_field_set(dialog, peersecret, peer->secret);
        ast_string_field_set(dialog, peermd5secret, peer->md5secret);
        ast_string_field_set(dialog, mohsuggest, peer->mohsuggest);
        ast_string_field_set(dialog, mohinterpret, peer->mohinterpret);
        ast_string_field_set(dialog, tohost, peer->tohost);
        ast_string_field_set(dialog, fullcontact, peer->fullcontact);
        if (!dialog->initreq.headers && !ast_strlen_zero(peer->fromdomain)) {
                char *tmpcall;
                char *c;
                tmpcall = ast_strdupa(dialog->callid);
                c = strchr(tmpcall, '@');
                if (c) {
                        *c = '\0';
                        ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
                }
        }
        if (ast_strlen_zero(dialog->tohost))
                ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
        if (!ast_strlen_zero(peer->fromdomain))
                ast_string_field_set(dialog, fromdomain, peer->fromdomain);
        if (!ast_strlen_zero(peer->fromuser))
                ast_string_field_set(dialog, fromuser, peer->fromuser);
        dialog->callgroup = peer->callgroup;
        dialog->pickupgroup = peer->pickupgroup;
        dialog->allowtransfer = peer->allowtransfer;
        /* Set timer T1 to RTT for this peer (if known by qualify=) */
        /* Minimum is settable or default to 100 ms */
        if (peer->maxms && peer->lastms)
                dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
        if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
            (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
                dialog->noncodeccapability |= AST_RTP_DTMF;
        else
                dialog->noncodeccapability &= ~AST_RTP_DTMF;
        ast_string_field_set(dialog, context, peer->context);
        dialog->rtptimeout = peer->rtptimeout;
        if (peer->call_limit)
                ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
        dialog->maxcallbitrate = peer->maxcallbitrate;
        
        return 0;
}

/*! \brief create address structure from peer name
 *      Or, if peer not found, find it in the global DNS 
 *      returns TRUE (-1) on failure, FALSE on success */
static int create_addr(struct sip_pvt *dialog, const char *opeer)
{
        struct hostent *hp;
        struct ast_hostent ahp;
        struct sip_peer *peer;
        char *port;
        int portno;
        char host[MAXHOSTNAMELEN], *hostn;
        char peername[256];

        ast_copy_string(peername, opeer, sizeof(peername));
        port = strchr(peername, ':');
        if (port)
                *port++ = '\0';
        dialog->sa.sin_family = AF_INET;
        dialog->timer_t1 = SIP_TIMER_T1; /* Default SIP retransmission timer T1 (RFC 3261) */
        peer = find_peer(peername, NULL, 1);

        if (peer) {
                int res = create_addr_from_peer(dialog, peer);
                unref_peer(peer);
                return res;
        }
        hostn = peername;
        portno = port ? atoi(port) : STANDARD_SIP_PORT;
        if (global_srvlookup) {
                char service[MAXHOSTNAMELEN];
                int tportno;
                int ret;

                snprintf(service, sizeof(service), "_sip._udp.%s", peername);
                ret = ast_get_srv(NULL, host, sizeof(host), &tportno, service);
                if (ret > 0) {
                        hostn = host;
                        portno = tportno;
                }
        }
        hp = ast_gethostbyname(hostn, &ahp);
        if (!hp) {
                ast_log(LOG_WARNING, "No such host: %s\n", peername);
                return -1;
        }
        ast_string_field_set(dialog, tohost, peername);
        memcpy(&dialog->sa.sin_addr, hp->h_addr, sizeof(dialog->sa.sin_addr));
        dialog->sa.sin_port = htons(portno);
        dialog->recv = dialog->sa;
        return 0;
}

/*! \brief Scheduled congestion on a call */
static int auto_congest(void *nothing)
{
        struct sip_pvt *p = nothing;

        sip_pvt_lock(p);
        p->initid = -1;
        if (p->owner) {
                /* XXX fails on possible deadlock */
                if (!ast_channel_trylock(p->owner)) {
                        ast_log(LOG_NOTICE, "Auto-congesting %s\n", p->owner->name);
                        append_history(p, "Cong", "Auto-congesting (timer)");
                        ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
                        ast_channel_unlock(p->owner);
                }
        }
        sip_pvt_unlock(p);
        return 0;
}


/*! \brief Initiate SIP call from PBX 
 *      used from the dial() application      */
static int sip_call(struct ast_channel *ast, char *dest, int timeout)
{
        int res;
        struct sip_pvt *p;
        struct varshead *headp;
        struct ast_var_t *current;
        const char *referer = NULL;   /* SIP referrer */        

        p = ast->tech_pvt;
        if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
                ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
                return -1;
        }

        /* Check whether there is vxml_url, distinctive ring variables */
        headp=&ast->varshead;
        AST_LIST_TRAVERSE(headp,current,entries) {
                /* Check whether there is a VXML_URL variable */
                if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
                        p->options->vxml_url = ast_var_value(current);
                } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
                        p->options->uri_options = ast_var_value(current);
                } else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
                        /* Check whether there is a variable with a name starting with SIPADDHEADER */
                        p->options->addsipheaders = 1;
                } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
                        /* This is a transfered call */
                        p->options->transfer = 1;
                } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
                        /* This is the referrer */
                        referer = ast_var_value(current);
                } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
                        /* We're replacing a call. */
                        p->options->replaces = ast_var_value(current);
                } else if (!strcasecmp(ast_var_name(current), "T38CALL")) {
                        p->t38.state = T38_LOCAL_DIRECT;
                        if (option_debug)
                                ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
                }

        }
        
        res = 0;
        ast_set_flag(&p->flags[0], SIP_OUTGOING);

        if (p->options->transfer) {
                char buf[BUFSIZ/2];

                if (referer) {
                        if (sipdebug && option_debug > 2)
                                ast_log(LOG_DEBUG, "Call for %s transfered by %s\n", p->username, referer);
                        snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
                } else 
                        snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
                ast_string_field_set(p, cid_name, buf);
        } 
        if (option_debug)
                ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username);

        res = update_call_counter(p, INC_CALL_RINGING);

        if (res == -1)
                return res;

        p->callingpres = ast->cid.cid_pres;
        p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
        
        /* If there are no audio formats left to offer, punt */
        if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
                ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
                res = -1;
        } else {
                p->t38.jointcapability = p->t38.capability;
                if (option_debug > 1)
                        ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
                transmit_invite(p, SIP_INVITE, 1, 2);
                p->invitestate = INV_CALLING;
                
                /* Initialize auto-congest time */
                p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
        }

        return res;
}

/*! \brief Destroy registry object
        Objects created with the register= statement in static configuration */
static void sip_registry_destroy(struct sip_registry *reg)
{
        /* Really delete */
        if (option_debug > 2)
                ast_log(LOG_DEBUG, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname);

        if (reg->call) {
                /* Clear registry before destroying to ensure
                   we don't get reentered trying to grab the registry lock */
                reg->call->registry = NULL;
                if (option_debug > 2)
                        ast_log(LOG_DEBUG, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname);
                sip_destroy(reg->call);
        }
        if (reg->expire > -1)
                ast_sched_del(sched, reg->expire);
        if (reg->timeout > -1)
                ast_sched_del(sched, reg->timeout);
        ast_string_field_free_pools(reg);
        regobjs--;
        free(reg);
        
}

/*! \brief Execute destruction of SIP dialog structure, release memory */
static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
{
        struct sip_pvt *cur, *prev = NULL;
        struct sip_pkt *cp;

        if (sip_debug_test_pvt(p) || option_debug > 2)
                ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);

        /* Remove link from peer to subscription of MWI */
        if (p->relatedpeer && p->relatedpeer->mwipvt) 
                p->relatedpeer->mwipvt = NULL;

        if (dumphistory)
                sip_dump_history(p);

        if (p->options)
                free(p->options);

        if (p->stateid > -1)
                ast_extension_state_del(p->stateid, NULL);
        if (p->initid > -1)
                ast_sched_del(sched, p->initid);
        if (p->autokillid > -1)
                ast_sched_del(sched, p->autokillid);

        if (p->rtp)
                ast_rtp_destroy(p->rtp);
        if (p->vrtp)
                ast_rtp_destroy(p->vrtp);
        if (p->udptl)
                ast_udptl_destroy(p->udptl);
        if (p->refer)
                free(p->refer);
        if (p->route) {
                free_old_route(p->route);
                p->route = NULL;
        }
        if (p->registry) {
                if (p->registry->call == p)
                        p->registry->call = NULL;
                unref_registry(p->registry);
        }

        /* Unlink us from the owner if we have one */
        if (p->owner) {
                if (lockowner)
                        ast_channel_lock(p->owner);
                if (option_debug)
                        ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
                p->owner->tech_pvt = NULL;
                if (lockowner)
                        ast_channel_unlock(p->owner);
        }
        /* Clear history */
        if (p->history) {
                struct sip_history *hist;
                while( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) )
                        free(hist);
                free(p->history);
                p->history = NULL;
        }

        /* Lock dialog list before removing ourselves from the list */
        if (lockdialoglist)
                dialoglist_lock();
        for (prev = NULL, cur = dialoglist; cur; prev = cur, cur = cur->next) {
                if (cur == p) {
                        UNLINK(cur, dialoglist, prev);
                        break;
                }
        }
        if (lockdialoglist)
                dialoglist_unlock();
        if (!cur) {
                ast_log(LOG_WARNING, "Trying to destroy \"%s\", not found in dialog list?!?! \n", p->callid);
                return;
        } 

        /* remove all current packets in this dialog */
        while((cp = p->packets)) {
                p->packets = p->packets->next;
                if (cp->retransid > -1)
                        ast_sched_del(sched, cp->retransid);
                free(cp);
        }
        if (p->chanvars) {
                ast_variables_destroy(p->chanvars);
                p->chanvars = NULL;
        }
        ast_mutex_destroy(&p->pvt_lock);

        ast_string_field_free_pools(p);

        free(p);
}

/*! \brief  update_call_counter: Handle call_limit for SIP users 
 * Setting a call-limit will cause calls above the limit not to be accepted.
 *
 * Remember that for a type=friend, there's one limit for the user and
 * another for the peer, not a combined call limit.
 * This will cause unexpected behaviour in subscriptions, since a "friend"
 * is *two* devices in Asterisk, not one.
 *
 * Thought: For realtime, we should probably update storage with inuse counter... 
 *
 * \return 0 if call is ok (no call limit, below threshold)
 *      -1 on rejection of call
 *              
 */
static int update_call_counter(struct sip_pvt *fup, int event)
{
        char name[256];
        int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
        int outgoing = ast_test_flag(&fup->flags[0], SIP_OUTGOING);
        struct sip_user *u = NULL;
        struct sip_peer *p = NULL;

        if (option_debug > 2)
                ast_log(LOG_DEBUG, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
        /* Test if we need to check call limits, in order to avoid 
           realtime lookups if we do not need it */
        if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT))
                return 0;

        ast_copy_string(name, fup->username, sizeof(name));

        /* Check the list of users only for incoming calls */
        if (global_limitonpeers == FALSE && !outgoing && (u = find_user(name, 1)))  {
                inuse = &u->inUse;
                call_limit = &u->call_limit;
                inringing = NULL;
        } else if ( (p = find_peer(ast_strlen_zero(fup->peername) ? name : fup->peername, NULL, 1) ) ) { /* Try to find peer */
                inuse = &p->inUse;
                call_limit = &p->call_limit;
                inringing = &p->inRinging;
                ast_copy_string(name, fup->peername, sizeof(name));
        } 
        if (!p && !u) {
                if (option_debug > 1)
                        ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name);
                return 0;
        }

        switch(event) {
        /* incoming and outgoing affects the inUse counter */
        case DEC_CALL_LIMIT:
                if ( *inuse > 0 ) {
                        if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT))
                                (*inuse)--;
                } else {
                        *inuse = 0;
                }
                if (inringing) {
                        if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
                                if (*inringing > 0)
                                        (*inringing)--;
                                else
                                        ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", fup->peername);
                                ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
                        }
                }
                if (option_debug > 1 || sipdebug) {
                        ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
                }
                break;

        case INC_CALL_RINGING:
        case INC_CALL_LIMIT:
                if (*call_limit > 0 ) {
                        if (*inuse >= *call_limit) {
                                ast_log(LOG_ERROR, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
                                if (u)
                                        unref_user(u);
                                else
                                        unref_peer(p);
                                return -1; 
                        }
                }
                if (inringing && (event == INC_CALL_RINGING)) {
                        if (!ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
                                (*inringing)++;
                                ast_set_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
                        }
                }
                /* Continue */
                (*inuse)++;
                ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
                if (option_debug > 1 || sipdebug) {
                        ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
                }
                break;

        case DEC_CALL_RINGING:
                if (inringing) {
                        if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
                                if (*inringing > 0)
                                        (*inringing)--;
                                else
                                        ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", p->name);
                                ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
                        }
                }
                break;

        default:
                ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
        }
        if (p) {
                ast_device_state_changed("SIP/%s", p->name);
                unref_peer(p);
        } else /* u must be set */
                unref_user(u);
        return 0;
}

/*! \brief Destroy SIP call structure */
static void sip_destroy(struct sip_pvt *p)
{
        if (option_debug > 2)
                ast_log(LOG_DEBUG, "Destroying SIP dialog %s\n", p->callid);
        __sip_destroy(p, TRUE, TRUE);
}

/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
static int hangup_sip2cause(int cause)
{
        /* Possible values taken from causes.h */

        switch(cause) {
                case 401:       /* Unauthorized */
                        return AST_CAUSE_CALL_REJECTED;
                case 403:       /* Not found */
                        return AST_CAUSE_CALL_REJECTED;
                case 404:       /* Not found */
                        return AST_CAUSE_UNALLOCATED;
                case 405:       /* Method not allowed */
                        return AST_CAUSE_INTERWORKING;
                case 407:       /* Proxy authentication required */
                        return AST_CAUSE_CALL_REJECTED;
                case 408:       /* No reaction */
                        return AST_CAUSE_NO_USER_RESPONSE;
                case 409:       /* Conflict */
                        return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
                case 410:       /* Gone */
                        return AST_CAUSE_UNALLOCATED;
                case 411:       /* Length required */
                        return AST_CAUSE_INTERWORKING;
                case 413:       /* Request entity too large */
                        return AST_CAUSE_INTERWORKING;
                case 414:       /* Request URI too large */
                        return AST_CAUSE_INTERWORKING;
                case 415:       /* Unsupported media type */
                        return AST_CAUSE_INTERWORKING;
                case 420:       /* Bad extension */
                        return AST_CAUSE_NO_ROUTE_DESTINATION;
                case 480:       /* No answer */
                        return AST_CAUSE_NO_ANSWER;
                case 481:       /* No answer */
                        return AST_CAUSE_INTERWORKING;
                case 482:       /* Loop detected */
                        return AST_CAUSE_INTERWORKING;
                case 483:       /* Too many hops */
                        return AST_CAUSE_NO_ANSWER;
                case 484:       /* Address incomplete */
                        return AST_CAUSE_INVALID_NUMBER_FORMAT;
                case 485:       /* Ambiguous */
                        return AST_CAUSE_UNALLOCATED;
                case 486:       /* Busy everywhere */
                        return AST_CAUSE_BUSY;
                case 487:       /* Request terminated */
                        return AST_CAUSE_INTERWORKING;
                case 488:       /* No codecs approved */
                        return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
                case 491:       /* Request pending */
                        return AST_CAUSE_INTERWORKING;
                case 493:       /* Undecipherable */
                        return AST_CAUSE_INTERWORKING;
                case 500:       /* Server internal failure */
                        return AST_CAUSE_FAILURE;
                case 501:       /* Call rejected */
                        return AST_CAUSE_FACILITY_REJECTED;
                case 502:       
                        return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
                case 503:       /* Service unavailable */
                        return AST_CAUSE_CONGESTION;
                case 504:       /* Gateway timeout */
                        return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
                case 505:       /* SIP version not supported */
                        return AST_CAUSE_INTERWORKING;
                case 600:       /* Busy everywhere */
                        return AST_CAUSE_USER_BUSY;
                case 603:       /* Decline */
                        return AST_CAUSE_CALL_REJECTED;
                case 604:       /* Does not exist anywhere */
                        return AST_CAUSE_UNALLOCATED;
                case 606:       /* Not acceptable */
                        return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
                default:
                        return AST_CAUSE_NORMAL;
        }
        /* Never reached */
        return 0;
}

/*! \brief Convert Asterisk hangup causes to SIP codes 
\verbatim
 Possible values from causes.h
        AST_CAUSE_NOTDEFINED    AST_CAUSE_NORMAL        AST_CAUSE_BUSY
        AST_CAUSE_FAILURE       AST_CAUSE_CONGESTION    AST_CAUSE_UNALLOCATED

        In addition to these, a lot of PRI codes is defined in causes.h 
        ...should we take care of them too ?
        
        Quote RFC 3398

   ISUP Cause value                        SIP response
   ----------------                        ------------
   1  unallocated number                   404 Not Found
   2  no route to network                  404 Not found
   3  no route to destination              404 Not found
   16 normal call clearing                 --- (*)
   17 user busy                            486 Busy here
   18 no user responding                   408 Request Timeout
   19 no answer from the user              480 Temporarily unavailable
   20 subscriber absent                    480 Temporarily unavailable
   21 call rejected                        403 Forbidden (+)
   22 number changed (w/o diagnostic)      410 Gone
   22 number changed (w/ diagnostic)       301 Moved Permanently
   23 redirection to new destination       410 Gone
   26 non-selected user clearing           404 Not Found (=)
   27 destination out of order             502 Bad Gateway
   28 address incomplete                   484 Address incomplete
   29 facility rejected                    501 Not implemented
   31 normal unspecified                   480 Temporarily unavailable
\endverbatim
*/
static const char *hangup_cause2sip(int cause)
{
        switch (cause) {
                case AST_CAUSE_UNALLOCATED:             /* 1 */
                case AST_CAUSE_NO_ROUTE_DESTINATION:    /* 3 IAX2: Can't find extension in context */
                case AST_CAUSE_NO_ROUTE_TRANSIT_NET:    /* 2 */
                        return "404 Not Found";
                case AST_CAUSE_CONGESTION:              /* 34 */
                case AST_CAUSE_SWITCH_CONGESTION:       /* 42 */
                        return "503 Service Unavailable";
                case AST_CAUSE_NO_USER_RESPONSE:        /* 18 */
                        return "408 Request Timeout";
                case AST_CAUSE_NO_ANSWER:               /* 19 */
                        return "480 Temporarily unavailable";
                case AST_CAUSE_CALL_REJECTED:           /* 21 */
                        return "403 Forbidden";
                case AST_CAUSE_NUMBER_CHANGED:          /* 22 */
                        return "410 Gone";
                case AST_CAUSE_NORMAL_UNSPECIFIED:      /* 31 */
                        return "480 Temporarily unavailable";
                case AST_CAUSE_INVALID_NUMBER_FORMAT:
                        return "484 Address incomplete";
                case AST_CAUSE_USER_BUSY:
                        return "486 Busy here";
                case AST_CAUSE_FAILURE:
                        return "500 Server internal failure";
                case AST_CAUSE_FACILITY_REJECTED:       /* 29 */
                        return "501 Not Implemented";
                case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
                        return "503 Service Unavailable";
                /* Used in chan_iax2 */
                case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
                        return "502 Bad Gateway";
                case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL:       /* Can't find codec to connect to host */
                        return "488 Not Acceptable Here";
                        
                case AST_CAUSE_NOTDEFINED:
                default:
                        if (option_debug)
                                ast_log(LOG_DEBUG, "AST hangup cause %d (no match found in SIP)\n", cause);
                        return NULL;
        }

        /* Never reached */
        return 0;
}


/*! \brief  sip_hangup: Hangup SIP call
 * Part of PBX interface, called from ast_hangup */
static int sip_hangup(struct ast_channel *ast)
{
        struct sip_pvt *p = ast->tech_pvt;
        int needcancel = FALSE;
        int needdestroy = 0;
        struct ast_channel *oldowner = ast;

        if (!p) {
                if (option_debug)
                        ast_log(LOG_DEBUG, "Asked to hangup channel that was not connected\n");
                return 0;
        }

        if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
                if (option_debug >3)
                        ast_log(LOG_DEBUG, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
                if (p->autokillid > -1)
                        sip_cancel_destroy(p);
                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);        /* Really hang up next time */
                ast_clear_flag(&p->flags[0], SIP_NEEDDESTROY);
                p->owner->tech_pvt = NULL;
                p->owner = NULL;  /* Owner will be gone after we return, so take it away */
                return 0;
        }
        if (option_debug) {
                if (ast_test_flag(ast, AST_FLAG_ZOMBIE) && p->refer && option_debug)
                        ast_log(LOG_DEBUG, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid);
                else  {
                        if (option_debug)
                                ast_log(LOG_DEBUG, "Hangup call %s, SIP callid %s)\n", ast->name, p->callid);
                }
        }
        if (option_debug && ast_test_flag(ast, AST_FLAG_ZOMBIE)) 
                ast_log(LOG_DEBUG, "Hanging up zombie call. Be scared.\n");

        sip_pvt_lock(p);
        if (option_debug && sipdebug)
                ast_log(LOG_DEBUG, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
        update_call_counter(p, DEC_CALL_LIMIT);

        /* Determine how to disconnect */
        if (p->owner != ast) {
                ast_log(LOG_WARNING, "Huh?  We aren't the owner? Can't hangup call.\n");
                sip_pvt_unlock(p);
                return 0;
        }
        /* If the call is not UP, we need to send CANCEL instead of BYE */
        /* In case of re-invites, the call might be UP even though we have an incomplete invite transaction */
        if (p->invitestate < INV_COMPLETED && p->owner->_state != AST_STATE_UP) {
                needcancel = TRUE;
                if (option_debug > 3)
                        ast_log(LOG_DEBUG, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
        }

        /* Disconnect */
        if (p->vad)
                ast_dsp_free(p->vad);

        p->owner = NULL;
        ast->tech_pvt = NULL;

        /* Do not destroy this pvt until we have timeout or
           get an answer to the BYE or INVITE/CANCEL 
           If we get no answer during retransmit period, drop the call anyway.
           (Sorry, mother-in-law, you can't deny a hangup by sending
           603 declined to BYE...)
        */
        if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE))
                needdestroy = 1;        /* Set destroy flag at end of this function */
        else if (p->invitestate != INV_CALLING)
                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);

        /* Start the process if it's not already started */
        if (!ast_test_flag(&p->flags[0], SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) {
                if (needcancel) {       /* Outgoing call, not up */
                        if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
                                /* stop retransmitting an INVITE that has not received a response */
                                __sip_pretend_ack(p);

                                /* if we can't send right now, mark it pending */
                                if (p->invitestate == INV_CALLING) {
                                        /* We can't send anything in CALLING state */
                                        ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
                                        /* Do we need a timer here if we don't hear from them at all? */
                                } else {
                                        /* Send a new request: CANCEL */
                                        transmit_request(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
                                        /* Actually don't destroy us yet, wait for the 487 on our original 
                                           INVITE, but do set an autodestruct just in case we never get it. */
                                        needdestroy = 0;
                                        sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                                }
                                if ( p->initid != -1 ) {
                                        /* channel still up - reverse dec of inUse counter
                                           only if the channel is not auto-congested */
                                        update_call_counter(p, INC_CALL_LIMIT);
                                }
                        } else {        /* Incoming call, not up */
                                const char *res;
                                if (ast->hangupcause && (res = hangup_cause2sip(ast->hangupcause)))