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[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, and experimental TCP and TLS support
 * Configuration file \link Config_sip sip.conf \endlink
 *
 * ********** IMPORTANT *
 * \note TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration
 *      settings, dialplan commands and dialplans apps/functions
 * See \ref sip_tcp_tls
 * 
 *
 * ******** General TODO:s
 * \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() for udp only. In tcp, packets are read by the tcp_helper thread.
 * sipsock_read() function parses the packet and matches an existing
 * dialog or starts a new SIP dialog.
 * 
 * sipsock_read sends the packet to handle_incoming(), that parses a bit more.
 * If it is a response to an outbound request, the packet is sent to handle_response().
 * If it is a request, handle_incoming() sends it to one of a list of functions
 * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
 * sipsock_read locks the ast_channel if it exists (an active call) and
 * unlocks it after we have processed the SIP message.
 *
 * A new INVITE is sent to handle_request_invite(), that will end up
 * starting a new channel in the PBX, the new channel after that executing
 * in a separate channel thread. This is an incoming "call".
 * When the call is answered, either by a bridged channel or the PBX itself
 * the sip_answer() function is called.
 *
 * The actual media - Video or Audio - is mostly handled by the RTP subsystem
 * in rtp.c 
 * 
 * \par Outbound calls
 * Outbound calls are set up by the PBX through the sip_request_call()
 * function. After that, they are activated by sip_call().
 * 
 * \par Hanging up
 * The PBX issues a hangup on both incoming and outgoing calls through
 * the sip_hangup() function
 */

/*!  
 * \page sip_tcp_tls SIP TCP and TLS support
 * 
 * \par tcpfixes TCP implementation changes needed
 * \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
 * \todo Save TCP/TLS sessions in registry
 *      If someone registers a SIPS uri, this forces us to set up a TLS connection back.
 * \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
 * \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
 *       The tcpbindaddr config option should only be used to open ADDITIONAL ports
 *       So we should propably go back to
 *              bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
 *                              if tlsenable=yes, open TLS port (provided we also have cert)
 *              tcpbindaddr = extra address for additional TCP connections
 *              tlsbindaddr = extra address for additional TCP/TLS connections
 *              udpbindaddr = extra address for additional UDP connections
 *                      These three options should take multiple IP/port pairs
 *      Note: Since opening additional listen sockets is a *new* feature we do not have today
 *              the XXXbindaddr options needs to be disabled until we have support for it
 *              
 * \todo re-evaluate the transport= setting in sip.conf. This is right now not well
 *      thought of. If a device in sip.conf contacts us via TCP, we should not switch transport,
 *      even if udp is the configured first transport.
 *      
 * \todo Be prepared for one outbound and another incoming socket per pvt. This applies
 *       specially to communication with other peers (proxies).
 * \todo We need to test TCP sessions with SIP proxies and in regards
 *       to the SIP outbound specs.
 * \todo transport=tls was deprecated in RFC3261 and should not be used at all. See section 22.2.2.
 *
 * \todo If the message is smaller than the given Content-length, the request should get a 400 Bad request
 *       message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)
 * \todo Since we have had multidomain support in Asterisk for quite a while, we need to support
 *       multiple domains in our TLS implementation, meaning one socket and one cert per domain
 * \todo Selection of transport for a request needs to be done after we've parsed all route headers,
 *       also considering outbound proxy options.
 *              First request: Outboundproxy, routes, (reg contact or URI. If URI doesn't have port:  DNS naptr, srv, AAA)
 *              Intermediate requests: Outboundproxy(only when forced), routes, contact/uri
 *      DNS naptr support is crucial. A SIP uri might lead to a TLS connection.
 *      Also note that due to outbound proxy settings, a SIPS uri might have to be sent on UDP (not to recommend though)
 * \todo Default transports are set to UDP, which cause the wrong behaviour when contacting remote
 *      devices directly from the dialplan. UDP is only a fallback if no other method works,
 *      in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the
 *      MTU (like INIVTE with video, audio and RTT)  TCP should be preferred.
 *
 *      When dialling unconfigured peers (with no port number)  or devices in external domains
 *      NAPTR records MUST be consulted to find configured transport. If they are not found,
 *      SRV records for both TCP and UDP should be checked. If there's a record for TCP, use that.
 *      If there's no record for TCP, then use UDP as a last resort. If there's no SRV records,
 *      \note this only applies if there's no outbound proxy configured for the session. If an outbound
 *      proxy is configured, these procedures might apply for locating the proxy and determining
 *      the transport to use for communication with the proxy.
 * \par Other bugs to fix ----
 * __set_address_from_contact(const char *fullcontact, struct sockaddr_in *sin, int tcp)
 *      - sets TLS port as default for all TCP connections, unless other port is given in contact.
 * parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
 *      - assumes that the contact the UA registers is using the same transport as the REGISTER request, which is 
 *        a bad guess.
 *      - Does not save any information about TCP/TLS connected devices, which is a severe BUG, as discussed on the mailing list.
 * get_destination(struct sip_pvt *p, struct sip_request *oreq)
 *      - Doesn't store the information that we got an incoming SIPS request in the channel, so that
 *        we can require a secure signalling path OUT of Asterisk (on SIP or IAX2). Possibly, the call should
 *        fail on in-secure signalling paths if there's no override in our configuration. At least, provide a
 *        channel variable in the dialplan.
 * get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req)
 *      - As above, if we have a SIPS: uri in the refer-to header
 *      - Does not check transport in refer_to uri.
 */

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

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

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

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

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

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

*/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

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

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

#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj.h"
/* 
   Uncomment the define below,  if you are having refcount related memory leaks.
   With this uncommented, this module will generate a file, /tmp/refs, which contains
   a history of the ao2_ref() calls. To be useful, all calls to ao2_* functions should
   be modified to ao2_t_* calls, and include a tag describing what is happening with 
   enough detail, to make pairing up a reference count increment with its corresponding decrement.
   The refcounter program in utils/ can be invaluable in highlighting objects that are not
   balanced, along with the complete history for that object.
   In normal operation, the macros defined will throw away the tags, so they do not 
   affect the speed of the program at all. They can be considered to be documentation.
*/
/* #define  REF_DEBUG 1 */
#include "asterisk/astobj2.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
#include "asterisk/linkedlists.h"
#include "asterisk/stringfields.h"
#include "asterisk/monitor.h"
#include "asterisk/netsock.h"
#include "asterisk/localtime.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/threadstorage.h"
#include "asterisk/translate.h"
#include "asterisk/ast_version.h"
#include "asterisk/event.h"
#include "asterisk/tcptls.h"

/*** DOCUMENTATION
        <application name="SIPDtmfMode" language="en_US">
                <synopsis>
                        Change the dtmfmode for a SIP call.
                </synopsis>
                <syntax>
                        <parameter name="mode" required="true">
                                <enumlist>
                                        <enum name="inband" />
                                        <enum name="info" />
                                        <enum name="rfc2833" />
                                </enumlist>
                        </parameter>
                </syntax>
                <description>
                        <para>Changes the dtmfmode for a SIP call.</para>
                </description>
        </application>
        <application name="SIPAddHeader" language="en_US">
                <synopsis>
                        Add a SIP header to the outbound call.
                </synopsis>
                <syntax argsep=":">
                        <parameter name="Header" required="true" />
                        <parameter name="Content" required="true" />
                </syntax>
                <description>
                        <para>Adds a header to a SIP call placed with DIAL.</para>
                        <para>Remember to use the X-header if you are adding non-standard SIP
                        headers, like <literal>X-Asterisk-Accountcode:</literal>. Use this with care.
                        Adding the wrong headers may jeopardize the SIP dialog.</para>
                        <para>Always returns <literal>0</literal>.</para>
                </description>
        </application>
        <function name="SIP_HEADER" language="en_US">
                <synopsis>
                        Gets the specified SIP header.
                </synopsis>
                <syntax>
                        <parameter name="name" required="true" />
                        <parameter name="number">
                                <para>If not specified, defaults to <literal>1</literal>.</para>
                        </parameter>
                </syntax>
                <description>
                        <para>Since there are several headers (such as Via) which can occur multiple
                        times, SIP_HEADER takes an optional second argument to specify which header with
                        that name to retrieve. Headers start at offset <literal>1</literal>.</para>
                </description>
        </function>
        <function name="SIPPEER" language="en_US">
                <synopsis>
                        Gets SIP peer information.
                </synopsis>
                <syntax>
                        <parameter name="peername" required="true" />
                        <parameter name="item">
                                <enumlist>
                                        <enum name="ip">
                                                <para>(default) The ip address.</para>
                                        </enum>
                                        <enum name="port">
                                                <para>The port number.</para>
                                        </enum>
                                        <enum name="mailbox">
                                                <para>The configured mailbox.</para>
                                        </enum>
                                        <enum name="context">
                                                <para>The configured context.</para>
                                        </enum>
                                        <enum name="expire">
                                                <para>The epoch time of the next expire.</para>
                                        </enum>
                                        <enum name="dynamic">
                                                <para>Is it dynamic? (yes/no).</para>
                                        </enum>
                                        <enum name="callerid_name">
                                                <para>The configured Caller ID name.</para>
                                        </enum>
                                        <enum name="callerid_num">
                                                <para>The configured Caller ID number.</para>
                                        </enum>
                                        <enum name="callgroup">
                                                <para>The configured Callgroup.</para>
                                        </enum>
                                        <enum name="pickupgroup">
                                                <para>The configured Pickupgroup.</para>
                                        </enum>
                                        <enum name="codecs">
                                                <para>The configured codecs.</para>
                                        </enum>
                                        <enum name="status">
                                                <para>Status (if qualify=yes).</para>
                                        </enum>
                                        <enum name="regexten">
                                                <para>Registration extension.</para>
                                        </enum>
                                        <enum name="limit">
                                                <para>Call limit (call-limit).</para>
                                        </enum>
                                        <enum name="busylevel">
                                                <para>Configured call level for signalling busy.</para>
                                        </enum>
                                        <enum name="curcalls">
                                                <para>Current amount of calls. Only available if call-limit is set.</para>
                                        </enum>
                                        <enum name="language">
                                                <para>Default language for peer.</para>
                                        </enum>
                                        <enum name="accountcode">
                                                <para>Account code for this peer.</para>
                                        </enum>
                                        <enum name="useragent">
                                                <para>Current user agent id for peer.</para>
                                        </enum>
                                        <enum name="chanvar[name]">
                                                <para>A channel variable configured with setvar for this peer.</para>
                                        </enum>
                                        <enum name="codec[x]">
                                                <para>Preferred codec index number <replaceable>x</replaceable> (beginning with zero).</para>
                                        </enum>
                                </enumlist>
                        </parameter>
                </syntax>
                <description />
        </function>
        <function name="SIPCHANINFO" language="en_US">
                <synopsis>
                        Gets the specified SIP parameter from the current channel.
                </synopsis>
                <syntax>
                        <parameter name="item" required="true">
                                <enumlist>
                                        <enum name="peerip">
                                                <para>The IP address of the peer.</para>
                                        </enum>
                                        <enum name="recvip">
                                                <para>The source IP address of the peer.</para>
                                        </enum>
                                        <enum name="from">
                                                <para>The URI from the <literal>From:</literal> header.</para>
                                        </enum>
                                        <enum name="uri">
                                                <para>The URI from the <literal>Contact:</literal> header.</para>
                                        </enum>
                                        <enum name="useragent">
                                                <para>The useragent.</para>
                                        </enum>
                                        <enum name="peername">
                                                <para>The name of the peer.</para>
                                        </enum>
                                        <enum name="t38passthrough">
                                                <para><literal>1</literal> if T38 is offered or enabled in this channel,
                                                otherwise <literal>0</literal>.</para>
                                        </enum>
                                </enumlist>
                        </parameter>
                </syntax>
                <description />
        </function>
        <function name="CHECKSIPDOMAIN" language="en_US">
                <synopsis>
                        Checks if domain is a local domain.
                </synopsis>
                <syntax>
                        <parameter name="domain" required="true" />
                </syntax>
                <description>
                        <para>This function checks if the <replaceable>domain</replaceable> in the argument is configured
                        as a local SIP domain that this Asterisk server is configured to handle.
                        Returns the domain name if it is locally handled, otherwise an empty string.
                        Check the <literal>domain=</literal> configuration in <filename>sip.conf</filename>.</para>
                </description>
        </function>
 ***/

#ifndef FALSE
#define FALSE    0
#endif

#ifndef TRUE
#define TRUE     1
#endif

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

/* Arguments for find_peer */
#define FINDALLDEVICES  FALSE
#define FINDONLYUSERS   TRUE


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

#define XMIT_ERROR              -2

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

/* #define VOCAL_DATA_HACK */

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

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

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 mwi_expiry = DEFAULT_MWI_EXPIRY;

#define CALLERID_UNKNOWN        "Unknown"

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

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

#define SIP_MAX_HEADERS              64               /*!< Max amount of SIP headers to read */
#define SIP_MAX_LINES                64               /*!< Max amount of lines in SIP attachment (like SDP) */
#define SIP_MIN_PACKET               1024             /*!< Initialize size of memory to allocate for packets */

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

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

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

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

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

#define RTP     1
#define NO_RTP  0

/*! \brief Authorization scheme for call transfers 

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


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

/*! \brief States for the INVITE transaction, not the dialog 
        \note this is for the INVITE that sets up the dialog
*/
enum invitestates {
        INV_NONE = 0,           /*!< No state at all, maybe not an INVITE dialog */
        INV_CALLING = 1,        /*!< Invite sent, no answer */
        INV_PROCEEDING = 2,     /*!< We got/sent 1xx message */
        INV_EARLY_MEDIA = 3,    /*!< We got 18x message with to-tag back */
        INV_COMPLETED = 4,      /*!< Got final response with error. Wait for ACK, then CONFIRMED */
        INV_CONFIRMED = 5,      /*!< Confirmed response - we've got an ack (Incoming calls only) */
        INV_TERMINATED = 6,     /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done 
                                     The only way out of this is a BYE from one side */
        INV_CANCELLED = 7,      /*!< Transaction cancelled by client or server in non-terminated state */
};

/*! \brief Readable descriptions of device states.
       \note Should be aligned to above table as index */
static const struct invstate2stringtable {
        const enum invitestates state;
        const char *desc;
} invitestate2string[] = {
        {INV_NONE,              "None"  },
        {INV_CALLING,           "Calling (Trying)"},
        {INV_PROCEEDING,        "Proceeding "},
        {INV_EARLY_MEDIA,       "Early media"},
        {INV_COMPLETED,         "Completed (done)"},
        {INV_CONFIRMED,         "Confirmed (up)"},
        {INV_TERMINATED,        "Done"},
        {INV_CANCELLED,         "Cancelled"}
};

/*! \brief When sending a SIP message, we can send with a few options, depending on
        type of SIP request. UNRELIABLE is moslty used for responses to repeated requests,
        where the original response would be sent RELIABLE in an INVITE transaction */
enum xmittype {
        XMIT_CRITICAL = 2,              /*!< Transmit critical SIP message reliably, with re-transmits.
                                              If it fails, it's critical and will cause a teardown of the session */
        XMIT_RELIABLE = 1,              /*!< Transmit SIP message reliably, with re-transmits */
        XMIT_UNRELIABLE = 0,            /*!< Transmit SIP message without bothering with re-transmits */
};

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

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

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


/*! \brief Authentication types - proxy or www authentication 
        \note Endpoints, like Asterisk, should always use WWW authentication to
        allow multiple authentications in the same call - to the proxy and
        to the end point.
*/
enum sip_auth_type {
        PROXY_AUTH = 407,
        WWW_AUTH = 401,
};

/*! \brief Authentication result from check_auth* functions */
enum check_auth_result {
        AUTH_DONT_KNOW = -100,  /*!< no result, need to check further */
                /* XXX maybe this is the same as AUTH_NOT_FOUND */

        AUTH_SUCCESSFUL = 0,
        AUTH_CHALLENGE_SENT = 1,
        AUTH_SECRET_FAILED = -1,
        AUTH_USERNAME_MISMATCH = -2,
        AUTH_NOT_FOUND = -3,    /*!< returned by register_verify */
        AUTH_FAKE_AUTH = -4,
        AUTH_UNKNOWN_DOMAIN = -5,
        AUTH_PEER_NOT_DYNAMIC = -6,
        AUTH_ACL_FAILED = -7,
        AUTH_BAD_TRANSPORT = -8,
};

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

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

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

        REG_STATE_REGISTERED,   /*!< Registered and done */

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

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

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

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

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

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

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

/*! \brief definition of a sip proxy server
 *
 * For outbound proxies, a sip_peer will contain a reference to a 
 * dynamically allocated instance of a sip_proxy. A sip_pvt may also
 * contain a reference to a peer's outboundproxy, or it may contain
 * a reference to the global_outboundproxy.
 */
struct sip_proxy {
        char name[MAXHOSTNAMELEN];      /*!< DNS name of domain/host or IP */
        struct sockaddr_in ip;          /*!< Currently used IP address and port */
        time_t last_dnsupdate;          /*!< When this was resolved */
        enum sip_transport transport;   
        int force;                      /*!< If it's an outbound proxy, Force use of this outbound proxy for all outbound requests */
        /* Room for a SRV record chain based on the name */
};

/*! \brief argument for the 'show channels|subscriptions' callback. */
struct __show_chan_arg { 
        int fd;
        int subscriptions;
        int numchans;   /* return value */
};


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

/*! \brief SIP Request methods known by Asterisk 

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

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

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

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

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

/*!  Define SIP option tags, used in Require: and Supported: headers 
        We need to be aware of these properties in the phones to use 
        the replace: header. We should not do that without knowing
        that the other end supports it... 
        This is nothing we can configure, we learn by the dialog
        Supported: header on the REGISTER (peer) or the INVITE
        (other devices)
        We are not using many of these today, but will in the future.
        This is documented in RFC 3261
*/
#define SUPPORTED               1
#define NOT_SUPPORTED           0

/* SIP options */
#define SIP_OPT_REPLACES        (1 << 0)
#define SIP_OPT_100REL          (1 << 1)
#define SIP_OPT_TIMER           (1 << 2)
#define SIP_OPT_EARLY_SESSION   (1 << 3)
#define SIP_OPT_JOIN            (1 << 4)
#define SIP_OPT_PATH            (1 << 5)
#define SIP_OPT_PREF            (1 << 6)
#define SIP_OPT_PRECONDITION    (1 << 7)
#define SIP_OPT_PRIVACY         (1 << 8)
#define SIP_OPT_SDP_ANAT        (1 << 9)
#define SIP_OPT_SEC_AGREE       (1 << 10)
#define SIP_OPT_EVENTLIST       (1 << 11)
#define SIP_OPT_GRUU            (1 << 12)
#define SIP_OPT_TARGET_DIALOG   (1 << 13)
#define SIP_OPT_NOREFERSUB      (1 << 14)
#define SIP_OPT_HISTINFO        (1 << 15)
#define SIP_OPT_RESPRIORITY     (1 << 16)
#define SIP_OPT_FROMCHANGE      (1 << 17)
#define SIP_OPT_RECLISTINV      (1 << 18)
#define SIP_OPT_RECLISTSUB      (1 << 19)
#define SIP_OPT_OUTBOUND        (1 << 20)
#define SIP_OPT_UNKNOWN         (1 << 21)


/*! \brief 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 */
        /* RFC3262: PRACK 100% reliability */
        { SIP_OPT_100REL,       NOT_SUPPORTED,  "100rel" },     
        /* RFC3959: SIP Early session support */
        { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" },
        /* SIMPLE events:  RFC4662 */
        { SIP_OPT_EVENTLIST,    NOT_SUPPORTED,  "eventlist" },
        /* RFC 4916- Connected line ID updates */
        { SIP_OPT_FROMCHANGE,   NOT_SUPPORTED,  "from-change" },
        /* GRUU: Globally Routable User Agent URI's */
        { SIP_OPT_GRUU,         NOT_SUPPORTED,  "gruu" },
        /* RFC4244 History info */
        { SIP_OPT_HISTINFO,     NOT_SUPPORTED,  "histinfo" },
        /* RFC3911: SIP Join header support */
        { SIP_OPT_JOIN,         NOT_SUPPORTED,  "join" },
        /* Disable the REFER subscription, RFC 4488 */
        { SIP_OPT_NOREFERSUB,   NOT_SUPPORTED,  "norefersub" },
        /* SIP outbound - the final NAT battle - draft-sip-outbound */
        { SIP_OPT_OUTBOUND,     NOT_SUPPORTED,  "outbound" },
        /* 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" },
        /* RFC-ietf-sip-uri-list-conferencing-02.txt conference invite lists */
        { SIP_OPT_RECLISTINV,   NOT_SUPPORTED,  "recipient-list-invite" },
        /* RFC-ietf-sip-uri-list-subscribe-02.txt - subscription lists */
        { SIP_OPT_RECLISTSUB,   NOT_SUPPORTED,  "recipient-list-subscribe" },
        /* RFC3891: Replaces: header for transfer */
        { SIP_OPT_REPLACES,     SUPPORTED,      "replaces" },   
        /* One version of Polycom firmware has the wrong label */
        { SIP_OPT_REPLACES,     SUPPORTED,      "replace" },    
        /* RFC4412 Resource priorities */
        { SIP_OPT_RESPRIORITY,  NOT_SUPPORTED,  "resource-priority" },
        /* RFC3329: Security agreement mechanism */
        { SIP_OPT_SEC_AGREE,    NOT_SUPPORTED,  "sec_agree" },
        /* RFC4092: Usage of the SDP ANAT Semantics in the SIP */
        { SIP_OPT_SDP_ANAT,     NOT_SUPPORTED,  "sdp-anat" },
        /* RFC4028: SIP Session-Timers */
        { SIP_OPT_TIMER,        SUPPORTED,      "timer" },
        /* RFC4538: Target-dialog */
        { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED,  "tdialog" },
};


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

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

/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_SIP_PORT       5060
/*! \brief Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_TLS_PORT       5061

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

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

   These are default values in the source. There are other recommended values in the
   sip.conf.sample for new installations. These may differ to keep backwards compatibility,
   yet encouraging new behaviour on new installations 
 */
/*@{*/ 
#define DEFAULT_CONTEXT         "default"       /*!< The default context for [general] section as well as devices */
#define DEFAULT_MOHINTERPRET    "default"       /*!< The default music class */
#define DEFAULT_MOHSUGGEST      ""
#define DEFAULT_VMEXTEN         "asterisk"      /*!< Default voicemail extension */
#define DEFAULT_CALLERID        "asterisk"      /*!< Default caller ID */
#define DEFAULT_NOTIFYMIME      "application/simple-message-summary"
#define DEFAULT_ALLOWGUEST      TRUE
#define DEFAULT_RTPKEEPALIVE    0               /*!< Default RTPkeepalive setting */
#define DEFAULT_CALLCOUNTER     FALSE
#define DEFAULT_SRVLOOKUP       TRUE            /*!< Recommended setting is ON */
#define DEFAULT_COMPACTHEADERS  FALSE           /*!< Send compact (one-character) SIP headers. Default off */
#define DEFAULT_TOS_SIP         0               /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_AUDIO       0               /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_VIDEO       0               /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_TEXT        0               /*!< Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_COS_SIP         4               /*!< Level 2 class of service for SIP signalling */
#define DEFAULT_COS_AUDIO       5               /*!< Level 2 class of service for audio media  */
#define DEFAULT_COS_VIDEO       6               /*!< Level 2 class of service for video media */
#define DEFAULT_COS_TEXT        5               /*!< Level 2 class of service for text media (T.140) */
#define DEFAULT_ALLOW_EXT_DOM   TRUE            /*!< Allow external domains */
#define DEFAULT_REALM           "asterisk"      /*!< Realm for HTTP digest authentication */
#define DEFAULT_NOTIFYRINGING   TRUE            /*!< Notify devicestate system on ringing state */
#define DEFAULT_NOTIFYCID               DISABLED        /*!< Include CID with ringing notifications */
#define DEFAULT_PEDANTIC        FALSE           /*!< Avoid following SIP standards for dialog matching */
#define DEFAULT_AUTOCREATEPEER  FALSE           /*!< Don't create peers automagically */
#define DEFAULT_MATCHEXTERNIPLOCALLY FALSE      /*!< Match extern IP locally default setting */
#define DEFAULT_QUALIFY         FALSE           /*!< Don't monitor devices */
#define DEFAULT_CALLEVENTS      FALSE           /*!< Extra manager SIP call events */
#define DEFAULT_ALWAYSAUTHREJECT        FALSE   /*!< Don't reject authentication requests always */
#define DEFAULT_REGEXTENONQUALIFY FALSE
#define DEFAULT_T1MIN           100             /*!< 100 MS for minimal roundtrip time */
#define DEFAULT_MAX_CALL_BITRATE (384)          /*!< Max bitrate for video */
#ifndef DEFAULT_USERAGENT
#define DEFAULT_USERAGENT "Asterisk PBX"        /*!< Default Useragent: header unless re-defined in sip.conf */
#define DEFAULT_SDPSESSION "Asterisk PBX"       /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
#define DEFAULT_SDPOWNER "root"                 /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
#endif
/*@}*/ 

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

/*@}*/ 

/*! \name GlobalSettings
        Global settings apply to the channel (often settings you can change in the general section
        of sip.conf
*/
/*@{*/ 
/*! \brief a place to store all global settings for the sip channel driver 
*/
struct sip_settings {
        int peer_rtupdate;              /*!< G: Update database with registration data for peer? */
        int rtsave_sysname;             /*!< G: Save system name at registration? */
        int ignore_regexpire;           /*!< G: Ignore expiration of peer  */
        int rtautoclear;                /*!< Realtime ?? */
        int directrtpsetup;             /*!< Enable support for Direct RTP setup (no re-invites) */
        int pedanticsipchecking;        /*!< Extra checking ?  Default off */
        int autocreatepeer;             /*!< Auto creation of peers at registration? Default off. */
        int srvlookup;                  /*!< SRV Lookup on or off. Default is on */
        int allowguest;                 /*!< allow unauthenticated peers to connect? */
        int alwaysauthreject;           /*!< Send 401 Unauthorized for all failing requests */
        int compactheaders;             /*!< send compact sip headers */
        int allow_external_domains;     /*!< Accept calls to external SIP domains? */
        int callevents;                 /*!< Whether we send manager events or not */
        int regextenonqualify;          /*!< Whether to add/remove regexten when qualifying peers */
        int matchexterniplocally;       /*!< Match externip/externhost setting against localnet setting */
};

static struct sip_settings sip_cfg;

static int global_notifyringing;        /*!< Send notifications on ringing */
static int global_notifyhold;           /*!< Send notifications on hold */
static int global_match_auth_username;          /*!< Match auth username if available instead of From: Default off. */

static enum notifycid_setting global_notifycid; /*!< Send CID with ringing notifications */

static int global_relaxdtmf;            /*!< Relax DTMF */
static int global_rtptimeout;           /*!< Time out call if no RTP */
static int global_rtpholdtimeout;       /*!< Time out call if no RTP during hold */
static int global_rtpkeepalive;         /*!< Send RTP keepalives */
static int global_reg_timeout;  
static int global_regattempts_max;      /*!< Registration attempts before giving up */
static int global_callcounter;          /*!< Enable call counters for all devices. This is currently enabled by setting the peer
                                                call-limit to 999. When we remove the call-limit from the code, we can make it
                                                with just a boolean flag in the device structure */
static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */
static int global_allowsubscribe;       /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE 
                                            the global setting is in globals_flags[1] */
static unsigned int global_tos_sip;             /*!< IP type of service for SIP packets */
static unsigned int global_tos_audio;           /*!< IP type of service for audio RTP packets */
static unsigned int global_tos_video;           /*!< IP type of service for video RTP packets */
static unsigned int global_tos_text;            /*!< IP type of service for text RTP packets */
static unsigned int global_cos_sip;             /*!< 802.1p class of service for SIP packets */
static unsigned int global_cos_audio;           /*!< 802.1p class of service for audio RTP packets */
static unsigned int global_cos_video;           /*!< 802.1p class of service for video RTP packets */
static unsigned int global_cos_text;            /*!< 802.1p class of service for text RTP packets */
static int recordhistory;               /*!< Record SIP history. Off by default */
static int dumphistory;                 /*!< Dump history to verbose before destroying SIP dialog */
static char global_realm[MAXHOSTNAMELEN];               /*!< Default realm */
static char global_regcontext[AST_MAX_CONTEXT];         /*!< Context for auto-extensions */
static char global_useragent[AST_MAX_EXTENSION];        /*!< Useragent for the SIP channel */
static char global_sdpsession[AST_MAX_EXTENSION];       /*!< SDP session name for the SIP channel */
static char global_sdpowner[AST_MAX_EXTENSION]; /*!< SDP owner name for the SIP channel */
static int global_authfailureevents;            /*!< Whether we send authentication failure manager events or not. Default no. */
static int global_t1;                   /*!< T1 time */
static int global_t1min;                /*!< T1 roundtrip time minimum */
static int global_timer_b;                      /*!< Timer B - RFC 3261 Section 17.1.1.2 */
static int global_autoframing;                  /*!< Turn autoframing on or off. */
static struct sip_proxy global_outboundproxy;   /*!< Outbound proxy */
static int global_qualifyfreq;                  /*!< Qualify frequency */


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

static enum st_mode global_st_mode;           /*!< Mode of operation for Session-Timers           */
static enum st_refresher global_st_refresher; /*!< Session-Timer refresher                        */
static int global_min_se;                     /*!< Lowest threshold for session refresh interval  */
static int global_max_se;                     /*!< Highest threshold for session refresh interval */

/*@}*/ 

/*! \brief Global list of addresses dynamic peers are not allowed to use */
static struct ast_ha *global_contact_ha = NULL;
static int global_dynamic_exclude_static = 0;

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

static struct ast_flags global_flags[2] = {{0}};        /*!< global SIP_ flags */
static char used_context[AST_MAX_CONTEXT];              /*!< name of automatically created context for unloading */


AST_MUTEX_DEFINE_STATIC(netlock);

/*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
   when it's doing something critical. */
AST_MUTEX_DEFINE_STATIC(monlock);

AST_MUTEX_DEFINE_STATIC(sip_reload_lock);

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

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

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

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

/*! \brief The SIP socket definition */
struct sip_socket {
        enum sip_transport type;        /*!< UDP, TCP or TLS */
        int fd;                         /*!< Filed descriptor, the actual socket */
        uint16_t port;
        struct ast_tcptls_session_instance *ser;        /* If tcp or tls, a socket manager */
};

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

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

struct sip_pkt;

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

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

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

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

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


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

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

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

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

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

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

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

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

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

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

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

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

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

#define SIP_PAGE2_VIDEOSUPPORT          (1 << 14)       /*!< DP: Video supported if offered? */
#define SIP_PAGE2_TEXTSUPPORT           (1 << 15)       /*!< GDP: Global text enable */
#define SIP_PAGE2_ALLOWSUBSCRIBE        (1 << 16)       /*!< GP: Allow subscriptions from this peer? */
#define SIP_PAGE2_ALLOWOVERLAP          (1 << 17)       /*!< DP: Allow overlap dialing ? */
#define SIP_PAGE2_SUBSCRIBEMWIONLY      (1 << 18)       /*!< GP: Only issue MWI notification if subscribed to */

#define SIP_PAGE2_T38SUPPORT            (7 << 20)       /*!< GDP: T38 Fax Passthrough Support */
#define SIP_PAGE2_T38SUPPORT_UDPTL      (1 << 20)       /*!< GDP: T38 Fax Passthrough Support */
#define SIP_PAGE2_T38SUPPORT_RTP        (2 << 20)       /*!< GDP: T38 Fax Passthrough Support (not implemented) */
#define SIP_PAGE2_T38SUPPORT_TCP        (4 << 20)       /*!< GDP: T38 Fax Passthrough Support (not implemented) */

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

#define SIP_PAGE2_RFC2833_COMPENSATE    (1 << 25)       /*!< DP: Compensate for buggy RFC2833 implementations */
#define SIP_PAGE2_BUGGY_MWI             (1 << 26)       /*!< DP: Buggy CISCO MWI fix */
#define SIP_PAGE2_DIALOG_ESTABLISHED    (1 << 27)       /*!< 29: Has a dialog been established? */
#define SIP_PAGE2_FAX_DETECT            (1 << 28)               /*!< DP: Fax Detection support */
#define SIP_PAGE2_REGISTERTRYING        (1 << 29)       /*!< DP: Send 100 Trying on REGISTER attempts */
#define SIP_PAGE2_UDPTL_DESTINATION     (1 << 30)       /*!< DP: Use source IP of RTP as destination if NAT is enabled */
#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS   (1 << 31)       /*!< DP: Always set up video, even if endpoints don't support it */

#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_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | SIP_PAGE2_UDPTL_DESTINATION | \
        SIP_PAGE2_VIDEOSUPPORT_ALWAYS)

/*@}*/ 

/*! \name SIPflagsT38
        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;
/*@}*/ 

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

static enum sip_debug_e sipdebug;

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

/*! \brief T38 States for a call */
enum t38state {
        T38_DISABLED = 0,                /*!< Not enabled */
        T38_LOCAL_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 (unused) */
        REFER_ACCEPTED,                /*!< Accepted by transferee */
        REFER_RINGING,                 /*!< Target Ringing */
        REFER_200OK,                   /*!< Answered by transfer target */
        REFER_FAILED,                  /*!< REFER declined - go on */
        REFER_NOAUTH                   /*!< We had no auth for REFER */
};

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

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

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


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


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




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

        /* boolean or small integers that don't belong in flags */
        char do_history;                        /*!< Set if we want to record history */
        char alreadygone;                       /*!< already destroyed by our peer */
        char needdestroy;                       /*!< need to be destroyed by the monitor thread */
        char outgoing_call;                     /*!< this is an outgoing call */
        char answered_elsewhere;                /*!< This call is cancelled due to answer on another channel */
        char novideo;                           /*!< Didn't get video in invite, don't offer */
        char notext;                            /*!< Text not supported  (?) */

        int timer_t1;                           /*!< SIP timer T1, ms rtt */
        int timer_b;                            /*!< SIP timer B, ms */
        unsigned int sipoptions;                /*!< Supported SIP options on the other end */
        unsigned int reqsipoptions;             /*!< Required SIP options on the other end */
        struct ast_codec_pref prefs;            /*!< codec prefs */
        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 jointnoncodeccapability;            /*!< Joint Non codec capability */
        int redircodecs;                        /*!< Redirect codecs */
        int maxcallbitrate;                     /*!< Maximum Call Bitrate for Video Calls */    
        struct sip_proxy *outboundproxy;        /*!< Outbound proxy for this dialog. Use ref_proxy to set this instead of setting it directly*/
        struct t38properties t38;               /*!< T38 settings */
        struct sockaddr_in udptlredirip;        /*!< Where our T.38 UDPTL should be going if not to us */
        struct ast_udptl *udptl;                /*!< T.38 UDPTL session */
        int callingpres;                        /*!< Calling presentation */
        int authtries;                          /*!< Times we've tried to authenticate */
        int expiry;                             /*!< How long we take to expire */
        long branch;                            /*!< The branch identifier of this session */
        long invite_branch;                     /*!< The branch used when we sent the initial INVITE */
        char tag[11];                           /*!< Our tag for this session */
        int sessionid;                          /*!< SDP Session ID */
        int sessionversion;                     /*!< SDP Session Version */
        uint64_t sessionversion_remote;         /*!< Remote UA's SDP Session Version */
        int session_modify;                     /*!< Session modification request true/false  */
        struct sockaddr_in sa;                  /*!< Our peer */
        struct sockaddr_in redirip;             /*!< Where our RTP should be going if not to us */
        struct sockaddr_in vredirip;            /*!< Where our Video RTP should be going if not to us */
        struct sockaddr_in tredirip;            /*!< Where our Text RTP should be going if not to us */
        time_t lastrtprx;                       /*!< Last RTP received */
        time_t lastrtptx;                       /*!< Last RTP sent */
        int rtptimeout;                         /*!< RTP timeout time */
        struct sockaddr_in recv;                /*!< Received as */
        struct sockaddr_in ourip;               /*!< Our IP (as seen from the outside) */
        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 ast_variable *notify_headers;    /*!< Custom notify type */      
        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 or state NOTIFY (in subscribe pvt's) ? (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 waitid;                             /*!< Wait ID for scheduler after 491 or other delays */
        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 ast_rtp *trtp;                   /*!< Text RTP session */
        struct sip_pkt *packets;                /*!< Packets scheduled for re-transmission */
        struct sip_history_head *history;       /*!< History of this SIP dialog */
        size_t history_entries;                 /*!< Number of entires in the history */
        struct ast_variable *chanvars;          /*!< Channel variables to set for inbound call */
        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) */
        struct sip_st_dlg *stimer;              /*!< SIP Session-Timers */              
  
        int red;                                /*!< T.140 RTP Redundancy */

        struct sip_subscription_mwi *mwi;       /*!< If this is a subscription MWI dialog, to which subscription */
};

/*! Max entires in the history list for a sip_pvt */
#define MAX_HISTORY_ENTRIES 50

/*! \brief
 * Here we implement the container for dialogs (sip_pvt), defining
 * generic wrapper functions to ease the transition from the current
 * implementation (a single linked list) to a different container.
 * In addition to a reference to the container, we need functions to lock/unlock
 * the container and individual items, and functions to add/remove
 * references to the individual items.
 */
struct ao2_container *dialogs;

#define sip_pvt_lock(x) ao2_lock(x)
#define sip_pvt_trylock(x) ao2_trylock(x)
#define sip_pvt_unlock(x) ao2_unlock(x)

/*! \brief
 * when we create or delete references, make sure to use these
 * functions so we keep track of the refcounts.
 * To simplify the code, we allow a NULL to be passed to dialog_unref().
 */
#ifdef REF_DEBUG
#define dialog_ref(arg1,arg2) dialog_ref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
#define dialog_unref(arg1,arg2) dialog_unref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)

static struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func)
{
        if (p)
                _ao2_ref_debug(p, 1, tag, file, line, func);
        else
                ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
        return p;
}

static struct sip_pvt *dialog_unref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func)
{
        if (p)
                _ao2_ref_debug(p, -1, tag, file, line, func);
        return NULL;
}
#else
static struct sip_pvt *dialog_ref(struct sip_pvt *p, char *tag)
{
        if (p)
                ao2_ref(p, 1);
        else
                ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
        return p;
}

static struct sip_pvt *dialog_unref(struct sip_pvt *p, char *tag)
{
        if (p)
                ao2_ref(p, -1);
        return NULL;
}
#endif

/*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission
 * Packets are linked in a list, whose head is in the struct sip_pvt they belong to.
 * Each packet holds a reference to the parent struct sip_pvt.
 * This structure is allocated in __sip_reliable_xmit() and only for packets that
 * require retransmissions.
 */
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 */
        char is_resp;                           /*!< 1 if this is a response packet (e.g. 200 OK), 0 if it is a request */
        char is_fatal;                          /*!< non-zero if there is a fatal error */
        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 */
        struct ast_str *data;
};      

/*!
 * \brief A peer's mailbox
 *
 * We could use STRINGFIELDS here, but for only two strings, it seems like
 * too much effort ...
 */
struct sip_mailbox {
        char *mailbox;
        char *context;
        /*! Associated MWI subscription */
        struct ast_event_sub *event_sub;
        AST_LIST_ENTRY(sip_mailbox) entry;
};

/*! \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, as will astobj2 hashing of the structure */
struct sip_peer {
        char name[80];                                  /*!< the unique name of this object */
        AST_DECLARE_STRING_FIELDS(
                AST_STRING_FIELD(secret);               /*!< Password for inbound auth */
                AST_STRING_FIELD(md5secret);            /*!< Password in MD5 */
                AST_STRING_FIELD(remotesecret);         /*!< Remote secret (trunks, remote devices) */
                AST_STRING_FIELD(context);              /*!< Default context for incoming calls */
                AST_STRING_FIELD(subscribecontext);     /*!< Default context for subscriptions */
                AST_STRING_FIELD(username);             /*!< Temporary username until registration */ 
                AST_STRING_FIELD(accountcode);          /*!< Account code */
                AST_STRING_FIELD(tohost);               /*!< If not dynamic, IP address */
                AST_STRING_FIELD(regexten);             /*!< Extension to register (if regcontext is used) */
                AST_STRING_FIELD(fromuser);             /*!< From: user when calling this peer */
                AST_STRING_FIELD(fromdomain);           /*!< From: domain when calling this peer */
                AST_STRING_FIELD(fullcontact);          /*!< Contact registered with us (not in sip.conf) */
                AST_STRING_FIELD(cid_num);              /*!< Caller ID num */
                AST_STRING_FIELD(cid_name);             /*!< Caller ID name */
                AST_STRING_FIELD(vmexten);              /*!< Dialplan extension for MWI notify message*/
                AST_STRING_FIELD(language);             /*!<  Default language for prompts */
                AST_STRING_FIELD(mohinterpret);         /*!<  Music on Hold class */
                AST_STRING_FIELD(mohsuggest);           /*!<  Music on Hold class */
                AST_STRING_FIELD(parkinglot);           /*!<  Parkinglot */
                AST_STRING_FIELD(useragent);            /*!<  User agent in SIP request (saved from registration) */
                );
        struct sip_socket socket;       /*!< Socket used for this peer */
        unsigned int transports:3;      /*!< Transports (enum sip_transport) that are acceptable for this peer */
        struct sip_auth *auth;          /*!< Realm authentication list */
        int amaflags;                   /*!< AMA Flags (for billing) */
        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 */
        int busy_level;                 /*!< Level of active channels where we signal busy */
        enum transfermodes allowtransfer;       /*! SIP Refer restriction scheme */
        struct ast_codec_pref prefs;    /*!<  codec prefs */
        int lastmsgssent;
        unsigned int sipoptions;        /*!<  Supported SIP options */
        struct ast_flags flags[2];      /*!<  SIP_ flags */
        
        /*! Mailboxes that this peer cares about */
        AST_LIST_HEAD_NOLOCK(, sip_mailbox) mailboxes;

        /* things that don't belong in flags */
        char is_realtime;               /*!< this is a 'realtime' peer */
        char rt_fromcontact;            /*!< P: copy fromcontact from realtime */
        char host_dynamic;              /*!< P: Dynamic Peers register with Asterisk */
        char selfdestruct;              /*!< P: Automatic peers need to destruct themselves */
        char onlymatchonip;             /*!< P: Only match on IP for incoming calls (old type=peer) */
        char the_mark;                  /*!< moved out of ASTOBJ into struct proper; That which bears the_mark should be deleted! */

        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 sip_proxy *outboundproxy;        /*!< Outbound proxy for this peer */
        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 */
        int qualifyfreq;                /*!<  Qualification: How often to check for the host to be up */
        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_ha *contactha;       /*!<  Restrict what IPs are allowed in the Contact header (for registration) */
        struct ast_variable *chanvars;  /*!<  Variables to set for channel created by user */
        struct sip_pvt *mwipvt;         /*!<  Subscription for MWI */
        int autoframing;
        struct sip_st_cfg stimer;       /*!<  SIP Session-Timers */
        int timer_t1;                   /*!<  The maximum T1 value for the peer */
        int timer_b;                    /*!<  The maximum timer B (transaction timeouts) */
        int deprecated_username; /*!< If it's a realtime peer, are they using the deprecated "username" instead of "defaultuser" */
};


/*! 
 * \brief Registrations with other SIP proxies
 *
 * Created by sip_register(), the entry is linked in the 'regl' list,
 * and never deleted (other than at 'sip reload' or module unload times).
 * The entry always has a pending timeout, either waiting for an ACK to
 * the REGISTER message (in which case we have to retransmit the request),
 * or waiting for the next REGISTER message to be sent (either the initial one,
 * or once the previously completed registration one expires).
 * The registration can be in one of many states, though at the moment
 * the handling is a bit mixed.
 *
 * XXX \todo Reference count handling for this object has some problems with
 * respect to scheduler entries.  The ref count is handled in some places,
 * but not all of them.  There are some places where references get leaked
 * when this scheduler entry gets cancelled.  At worst, this would cause
 * memory leaks on reloads if registrations get removed from configuration.
 */
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);
        );
        enum sip_transport transport;   /*!< Transport for this registration UDP, TCP or TLS */
        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) */
        struct timeval 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 ast_dnsmgr_entry *dnsmgr;        /*!<  DNS refresh manager for register */
        struct sockaddr_in us;          /*!< Who the server thinks we are */
        int noncecount;                 /*!< Nonce-count */
        char lastmsg[256];              /*!< Last Message sent/received */
};

/*! \brief Definition of a thread that handles a socket */
struct sip_threadinfo {
        int stop;
        pthread_t threadid;
        struct ast_tcptls_session_instance *ser;
        enum sip_transport type;        /*!< We keep a copy of the type here so we can display it in the connection list */
        AST_LIST_ENTRY(sip_threadinfo) list;
};

/*! \brief Definition of an MWI subscription to another server */
struct sip_subscription_mwi {
        ASTOBJ_COMPONENTS_FULL(struct sip_subscription_mwi,1,1);
        AST_DECLARE_STRING_FIELDS(
                AST_STRING_FIELD(username);     /*!< Who we are sending the subscription as */
                AST_STRING_FIELD(authuser);     /*!< Who we *authenticate* as */
                AST_STRING_FIELD(hostname);     /*!< Domain or host we subscribe to */
                AST_STRING_FIELD(secret);       /*!< Password in clear text */
                AST_STRING_FIELD(mailbox);      /*!< Mailbox store to put MWI into */
                );
        enum sip_transport transport;    /*!< Transport to use */
        int portno;                      /*!< Optional port override */
        int resub;                       /*!< Sched ID of resubscription */
        unsigned int subscribed:1;       /*!< Whether we are currently subscribed or not */
        struct sip_pvt *call;            /*!< Outbound subscription dialog */
        struct ast_dnsmgr_entry *dnsmgr; /*!< DNS refresh manager for subscription */
        struct sockaddr_in us;           /*!< Who the server thinks we are */
};

/* --- Hash tables of various objects --------*/

#ifdef LOW_MEMORY
static int hash_peer_size = 17;
static int hash_dialog_size = 17;
static int hash_user_size = 17;
#else
static int hash_peer_size = 563;        /*!< Size of peer hash table, prime number preferred! */
static int hash_dialog_size = 563;
static int hash_user_size = 563;
#endif

/*! \brief  The thread list of TCP threads */
static AST_LIST_HEAD_STATIC(threadl, sip_threadinfo);

/*! \brief  The peer list: Users, Peers and Friends */
struct ao2_container *peers;
struct ao2_container *peers_by_ip;

/*! \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;

/*! \brief  The MWI subscription list */
static struct ast_subscription_mwi_list {
        ASTOBJ_CONTAINER_COMPONENTS(struct sip_subscription_mwi);
} submwil;

/*! \brief
 * \note The only member of the peer used here is the name field
 */
static int peer_hash_cb(const void *obj, const int flags)
{
        const struct sip_peer *peer = obj;

        return ast_str_case_hash(peer->name);
}

/*!
 * \note The only member of the peer used here is the name field
 */
static int peer_cmp_cb(void *obj, void *arg, int flags)
{
        struct sip_peer *peer = obj, *peer2 = arg;

        return !strcasecmp(peer->name, peer2->name) ? CMP_MATCH | CMP_STOP : 0;
}

/*!
 * \note the peer's addr struct provides to fields combined to make a key: the sin_addr.s_addr and sin_port fields.
 */
static int peer_iphash_cb(const void *obj, const int flags)
{
        const struct sip_peer *peer = obj;
        int ret1 = peer->addr.sin_addr.s_addr;
        if (ret1 < 0)
                ret1 = -ret1;
        
        if (ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT)) {
                return ret1;
        } else {
                return ret1 + peer->addr.sin_port;
        }
}

/*!
 * \note the peer's addr struct provides to fields combined to make a key: the sin_addr.s_addr and sin_port fields.
 */
static int peer_ipcmp_cb(void *obj, void *arg, int flags)
{
        struct sip_peer *peer = obj, *peer2 = arg;

        if (peer->addr.sin_addr.s_addr != peer2->addr.sin_addr.s_addr)
                return 0;
        
        if (!ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT) && !ast_test_flag(&peer2->flags[0], SIP_INSECURE_PORT)) {
                if (peer->addr.sin_port == peer2->addr.sin_port)
                        return CMP_MATCH | CMP_STOP;
                else
                        return 0;
        }
        return CMP_MATCH | CMP_STOP;
}

/*!
 * \note The only member of the dialog used here callid string
 */
static int dialog_hash_cb(const void *obj, const int flags)
{
        const struct sip_pvt *pvt = obj;

        return ast_str_case_hash(pvt->callid);
}

/*!
 * \note The only member of the dialog used here callid string
 */
static int dialog_cmp_cb(void *obj, void *arg, int flags)
{
        struct sip_pvt *pvt = obj, *pvt2 = arg;
        
        return !strcasecmp(pvt->callid, pvt2->callid) ? CMP_MATCH | CMP_STOP : 0;
}

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);

#ifdef LOW_MEMORY
static void ts_ast_rtp_destroy(void *);

AST_THREADSTORAGE_CUSTOM(ts_audio_rtp, NULL, ts_ast_rtp_destroy);
AST_THREADSTORAGE_CUSTOM(ts_video_rtp, NULL, ts_ast_rtp_destroy);
AST_THREADSTORAGE_CUSTOM(ts_text_rtp, NULL, ts_ast_rtp_destroy);
#endif

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


/* --- Sockets and networking --------------*/

/*! \brief Main socket for UDP SIP communication.
 *
 * sipsock is shared between the SIP manager thread (which handles reload
 * requests), the udp io handler (sipsock_read()) and the user routines that
 * issue udp writes (using __sip_xmit()).
 * The socket is -1 only when opening fails (this is a permanent condition),
 * or when we are handling a reload() that changes its address (this is
 * a transient situation during which we might have a harmless race, see
 * below). Because the conditions for the race to be possible are extremely
 * rare, we don't want to pay the cost of locking on every I/O.
 * Rather, we remember that when the race may occur, communication is
 * bound to fail anyways, so we just live with this event and let
 * the protocol handle this above us.
 */
static int sipsock  = -1;

static struct sockaddr_in bindaddr;     /*!< UDP: The address we bind to */

/*! \brief our (internal) default address/port to put in SIP/SDP messages
 *  internip is initialized picking a suitable address from one of the
 * interfaces, and the same port number we bind to. It is used as the
 * default address/port in SIP messages, and as the default address
 * (but not port) in SDP messages.
 */
static struct sockaddr_in internip;

/*! \brief our external IP address/port for SIP sessions.
 * externip.sin_addr is only set when we know we might be behind
 * a NAT, and this is done using a variety of (mutually exclusive)
 * ways from the config file:
 *
 * + with "externip = host[:port]" we specify the address/port explicitly.
 *   The address is looked up only once when (re)loading the config file;
 * 
 * + with "externhost = host[:port]" we do a similar thing, but the
 *   hostname is stored in externhost, and the hostname->IP mapping
 *   is refreshed every 'externrefresh' seconds;
 * 
 * + with "stunaddr = host[:port]" we run queries every externrefresh seconds
 *   to the specified server, and store the result in externip.
 *
 * Other variables (externhost, externexpire, externrefresh) are used
 * to support the above functions.
 */
static struct sockaddr_in externip;             /*!< External IP address if we are behind NAT */

static char externhost[MAXHOSTNAMELEN];         /*!< External host name */
static time_t externexpire;                     /*!< Expiration counter for re-resolving external host name in dynamic DNS */
static int externrefresh = 10;
static struct sockaddr_in stunaddr;             /*!< stun server address */

/*! \brief  List of local networks
 * We store "localnet" addresses from the config file into an access list,
 * marked as 'DENY', so the call to ast_apply_ha() will return
 * AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local'
 * (i.e. presumably public) addresses.
 */
static struct ast_ha *localaddr;                /*!< List of local networks, on the same side of NAT as this Asterisk */

static int ourport_tcp;                         /*!< The port used for TCP connections */
static int ourport_tls;                         /*!< The port used for TCP/TLS connections */
static struct sockaddr_in debugaddr;

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

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

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

/*---------------------------- 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_sendhtml(struct ast_channel *chan, int subclass, const char *data, int datalen);
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, unsigned int duration);
static int sip_queryoption(struct ast_channel *chan, int option, void *data, int *datalen);
static const char *sip_get_callid(struct ast_channel *chan);

static int handle_request_do(struct sip_request *req, struct sockaddr_in *sin);
static int sip_standard_port(enum sip_transport type, int port);
static int sip_prepare_socket(struct sip_pvt *p);
static int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport);

/*--- Transmitting responses and requests */
static int sipsock_read(int *id, int fd, short events, void *ignore);
static int __sip_xmit(struct sip_pvt *p, struct ast_str *data, int len);
static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, struct ast_str *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(const void *data);
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, int oldsdp);
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, int oldsdp);
static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration);
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, const char *vmexten);
static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
static int transmit_notify_custom(struct sip_pvt *p, struct ast_variable *vars);
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, const struct ast_event *event, int cache_only);

/*--- 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(const void *data);
static void sip_scheddestroy(struct sip_pvt *p, int ms);
static int sip_cancel_destroy(struct sip_pvt *p);
static struct sip_pvt *sip_destroy(struct sip_pvt *p);
static void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist);
static void *registry_unref(struct sip_registry *reg, char *tag);
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(const void *arg);
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, int t38action);
static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
                             struct ast_str **m_buf, struct ast_str **a_buf,
                             int debug, int *min_packet_size);
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
                                struct ast_str **m_buf, struct ast_str **a_buf,
                                int debug);
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp);
static void do_setnat(struct sip_pvt *p, int natflags);
static void stop_media_flows(struct sip_pvt *p);

/*--- 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, const 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 void check_rtp_timeout(struct sip_pvt *dialog, time_t t);
static int sip_do_reload(enum channelreloadreason reason);
static int reload_config(enum channelreloadreason reason);
static int expire_register(const void *data);
static void *do_monitor(void *data);
static int restart_monitor(void);
static void peer_mailboxes_to_str(struct ast_str **mailbox_str, struct sip_peer *peer);
/* 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);
static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);

/*!
 * \brief generic function for determining if a correct transport is being 
 * used to contact a peer
 *
 * this is done as a macro so that the "tmpl" var can be passed either a 
 * sip_request or a sip_peer 
 */
#define check_request_transport(peer, tmpl) ({ \
        int ret = 0; \
        if (peer->socket.type == tmpl->socket.type) \
                ; \
        else if (!(peer->transports & tmpl->socket.type)) {\
                ast_log(LOG_ERROR, \
                        "'%s' is not a valid transport for '%s'. we only use '%s'! ending call.\n", \
                        get_transport(tmpl->socket.type), peer->name, get_transport_list(peer->transports) \
                        ); \
                ret = 1; \
        } else if (peer->socket.type & SIP_TRANSPORT_TLS) { \
                ast_log(LOG_WARNING, \
                        "peer '%s' HAS NOT USED (OR SWITCHED TO) TLS in favor of '%s' (but this was allowed in sip.conf)!\n", \
                        peer->name, get_transport(tmpl->socket.type) \
                ); \
        } else { \
                ast_debug(1, \
                        "peer '%s' has contacted us over %s even though we prefer %s.\n", \
                        peer->name, get_transport(tmpl->socket.type), get_transport(peer->socket.type) \
                ); \
        }\
        (ret); \
})


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

/*--- Applications, functions, CLI and manager command helpers */
static const char *sip_nat_mode(const struct sip_pvt *p);
static char *sip_show_inuse(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *transfermode2str(enum transfermodes mode) attribute_const;
static const char *nat2str(int nat) attribute_const;
static int peer_status(struct sip_peer *peer, char *status, int statuslen);
static char *sip_show_sched(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char * _sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static char *sip_show_peers(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_objects(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static void  print_group(int fd, ast_group_t group, int crlf);
static const char *dtmfmode2str(int mode) attribute_const;
static int str2dtmfmode(const char *str) attribute_unused;
static const char *insecure2str(int mode) 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 char *sip_show_domains(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static char *sip_show_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *_sip_qualify_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static char *sip_qualify_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_registry(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_mwi(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
static char *complete_sip_peer(const char *word, int state, int flags2);
static char *complete_sip_registered_peer(const char *word, int state, int flags2);
static char *complete_sip_show_history(const char *line, const char *word, int pos, int state);
static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
static char *complete_sip_unregister(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 *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_do_debug_ip(int fd, char *arg);
static char *sip_do_debug_peer(int fd, char *arg);
static char *sip_do_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_cli_notify(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_set_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
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 char *sip_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static int acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen);

/*--- 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 debuglog 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);


/*! \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, ...);
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, int ispeer);
static int update_call_counter(struct sip_pvt *fup, int event);
static void sip_destroy_peer(struct sip_peer *peer);
static void sip_destroy_peer_fn(void *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, int forcenamematch, int devstate_only);
static int sip_poke_peer_s(const 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 void set_insecure_flags(struct ast_flags *flags, const char *value, int lineno);
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, const char *useragent, int expirey, int deprecated_username);
static void update_peer(struct sip_peer *p, int expire);
static struct ast_variable *get_insecure_variable_from_config(struct ast_config *config);
static const char *get_name_from_variable(struct ast_variable *var, const char *newpeername);
static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin, int devstate_only);
static char *sip_prune_realtime(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);

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

/*--- 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 int 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 char *remove_uri_parameters(char *uri);
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, int addnewline);
static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout);

/*-- TCP connection handling ---*/
static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *ser);
static void *sip_tcp_worker_fn(void *);

/*--- 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, struct sockaddr_in *sin, int newdialog);
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, unsigned int duration, int mode);
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_incoming(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, int *nounlock);
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 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_notify(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 void handle_response_subscribe(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,  struct ast_rtp *trtp, 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 enum ast_rtp_get_result sip_get_trtp_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); 
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);
static void change_t38_state(struct sip_pvt *p, int state);

/*------ Session-Timers functions --------- */
static void proc_422_rsp(struct sip_pvt *p, struct sip_request *rsp);
static int  proc_session_timer(const void *vp);
static void stop_session_timer(struct sip_pvt *p);
static void start_session_timer(struct sip_pvt *p);
static void restart_session_timer(struct sip_pvt *p);
static const char *strefresher2str(enum st_refresher r);
static int parse_session_expires(const char *p_hdrval, int *const p_interval, enum st_refresher *const p_ref);
static int parse_minse(const char *p_hdrval, int *const p_interval);
static int st_get_se(struct sip_pvt *, int max);
static enum st_refresher st_get_refresher(struct sip_pvt *);
static enum st_mode st_get_mode(struct sip_pvt *);
static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p);

/*!--- SIP MWI Subscription support */
static int sip_subscribe_mwi(const char *value, int lineno);
static void sip_subscribe_mwi_destroy(struct sip_subscription_mwi *mwi);
static void sip_send_all_mwi_subscriptions(void);
static int sip_subscribe_mwi_do(const void *data);
static int __sip_subscribe_mwi_do(struct sip_subscription_mwi *mwi);

/*! \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_AUDIO_MASK,  /* all audio formats */
        .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
        .requester = sip_request_call,                  /* called with chan unlocked */
        .devicestate = sip_devicestate,                 /* called with chan unlocked (not chan-specific) */
        .call = sip_call,                       /* called with chan locked */
        .send_html = sip_sendhtml,
        .hangup = sip_hangup,                   /* called with chan locked */
        .answer = sip_answer,                   /* called with chan locked */
        .read = sip_read,                       /* called with chan locked */
        .write = sip_write,                     /* called with chan locked */
        .write_video = sip_write,               /* called with chan locked */
        .write_text = sip_write,
        .indicate = sip_indicate,               /* called with chan locked */
        .transfer = sip_transfer,               /* called with chan locked */
        .fixup = sip_fixup,                     /* called with chan locked */
        .send_digit_begin = sip_senddigit_begin,        /* called with chan unlocked */
        .send_digit_end = sip_senddigit_end,
        .bridge = ast_rtp_bridge,                       /* XXX chan unlocked ? */
        .early_bridge = ast_rtp_early_bridge,
        .send_text = sip_sendtext,              /* called with chan locked */
        .func_channel_read = acf_channel_read,
        .queryoption = sip_queryoption,
        .get_pvt_uniqueid = sip_get_callid,
};

/*! \brief This version of the sip channel tech has no send_digit_begin
 * callback so that the core knows that the channel does not want
 * DTMF BEGIN frames.
 * The struct is initialized just before registering the channel driver,
 * and is for use with channels using SIP INFO DTMF.
 */
static struct ast_channel_tech sip_tech_info;


/*! \brief Working TLS connection configuration */
static struct ast_tls_config sip_tls_cfg;

/*! \brief Default TLS connection configuration */
static struct ast_tls_config default_tls_cfg;

/*! \brief The TCP server definition */
static struct ast_tcptls_session_args sip_tcp_desc = {
        .accept_fd = -1,
        .master = AST_PTHREADT_NULL,
        .tls_cfg = NULL,
        .poll_timeout = -1,
        .name = "SIP TCP server",
        .accept_fn = ast_tcptls_server_root,
        .worker_fn = sip_tcp_worker_fn,
};

/*! \brief The TCP/TLS server definition */
static struct ast_tcptls_session_args sip_tls_desc = {
        .accept_fd = -1,
        .master = AST_PTHREADT_NULL,
        .tls_cfg = &sip_tls_cfg,
        .poll_timeout = -1,
        .name = "SIP TLS server",
        .accept_fn = ast_tcptls_server_root,
        .worker_fn = sip_tcp_worker_fn,
};

/* wrapper macro to tell whether t points to one of the sip_tech descriptors */
#define IS_SIP_TECH(t)  ((t) == &sip_tech || (t) == &sip_tech_info)

/*! \brief map from an integer value to a string.
 * If no match is found, return errorstring
 */
static const char *map_x_s(const struct _map_x_s *table, int x, const char *errorstring)
{
        const struct _map_x_s *cur;

        for (cur = table; cur->s; cur++)
                if (cur->x == x)
                        return cur->s;
        return errorstring;
}

/*! \brief map from a string to an integer value, case insensitive.
 * If no match is found, return errorvalue.
 */
static int map_s_x(const struct _map_x_s *table, const char *s, int errorvalue)
{
        const struct _map_x_s *cur;

        for (cur = table; cur->s; cur++)
                if (!strcasecmp(cur->s, s))
                        return cur->x;
        return errorvalue;
}


/*! \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,
        .get_trtp_info = sip_get_trtp_peer,
        .set_rtp_peer = sip_set_rtp_peer,
        .get_codec = sip_get_codec,
};


/*! \brief SIP TCP connection handler */
static void *sip_tcp_worker_fn(void *data)
{
        struct ast_tcptls_session_instance *ser = data;

        return _sip_tcp_helper_thread(NULL, ser);
}

/*! \brief SIP TCP thread management function 
        This function reads from the socket, parses the packet into a request
*/
static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *ser) 
{
        int res, cl;
        struct sip_request req = { 0, } , reqcpy = { 0, };
        struct sip_threadinfo *me;
        char buf[1024] = "";

        me = ast_calloc(1, sizeof(*me));

        if (!me)
                goto cleanup2;

        me->threadid = pthread_self();
        me->ser = ser;
        if (ser->ssl)
                me->type = SIP_TRANSPORT_TLS;
        else
                me->type = SIP_TRANSPORT_TCP;

        ast_debug(2, "Starting thread for %s server\n", ser->ssl ? "SSL" : "TCP");

        AST_LIST_LOCK(&threadl);
        AST_LIST_INSERT_TAIL(&threadl, me, list);
        AST_LIST_UNLOCK(&threadl);

        if (!(req.data = ast_str_create(SIP_MIN_PACKET)))
                goto cleanup;
        if (!(reqcpy.data = ast_str_create(SIP_MIN_PACKET)))
                goto cleanup;

        for (;;) {
                struct ast_str *str_save;

                str_save = req.data;
                memset(&req, 0, sizeof(req));
                req.data = str_save;
                ast_str_reset(req.data);

                str_save = reqcpy.data;
                memset(&reqcpy, 0, sizeof(reqcpy));
                reqcpy.data = str_save;
                ast_str_reset(reqcpy.data);

                req.socket.fd = ser->fd;
                if (ser->ssl) {
                        req.socket.type = SIP_TRANSPORT_TLS;
                        req.socket.port = htons(ourport_tls);
                } else {
                        req.socket.type = SIP_TRANSPORT_TCP;
                        req.socket.port = htons(ourport_tcp);
                }
                res = ast_wait_for_input(ser->fd, -1);
                if (res < 0) {
                        ast_debug(2, "SIP %s server :: ast_wait_for_input returned %d\n", ser->ssl ? "SSL": "TCP", res);
                        goto cleanup;
                }

                /* Read in headers one line at a time */
                while (req.len < 4 || strncmp((char *)&req.data->str + req.len - 4, "\r\n\r\n", 4)) {
                        ast_mutex_lock(&ser->lock);
                        if (!fgets(buf, sizeof(buf), ser->f)) {
                                ast_mutex_unlock(&ser->lock);
                                goto cleanup;
                        }
                        ast_mutex_unlock(&ser->lock);
                        if (me->stop) 
                                 goto cleanup;
                        ast_str_append(&req.data, 0, "%s", buf);
                        req.len = req.data->used;
                }
                copy_request(&reqcpy, &req);
                parse_request(&reqcpy);
                /* In order to know how much to read, we need the content-length header */
                if (sscanf(get_header(&reqcpy, "Content-Length"), "%d", &cl)) {
                        while (cl > 0) {
                                ast_mutex_lock(&ser->lock);
                                if (!fread(buf, (cl < sizeof(buf)) ? cl : sizeof(buf), 1, ser->f)) {
                                        ast_mutex_unlock(&ser->lock);
                                        goto cleanup;
                                }
                                ast_mutex_unlock(&ser->lock);
                                if (me->stop)
                                        goto cleanup;
                                cl -= strlen(buf);
                                ast_str_append(&req.data, 0, "%s", buf);
                                req.len = req.data->used;
                        }
                }
                /*! \todo XXX If there's no Content-Length or if the content-length and what
                                we receive is not the same - we should generate an error */

                req.socket.ser = ser;
                handle_request_do(&req, &ser->remote_address);
        }

cleanup:
        AST_LIST_LOCK(&threadl);
        AST_LIST_REMOVE(&threadl, me, list);
        AST_LIST_UNLOCK(&threadl);
        ast_free(me);
cleanup2:
        fclose(ser->f);
        ser->f = NULL;
        ser->fd = -1;
        if (reqcpy.data) {
                ast_free(reqcpy.data);
        }

        if (req.data) {
                ast_free(req.data);
                req.data = NULL;
        }

        ast_debug(2, "Shutting down thread for %s server\n", ser->ssl ? "SSL" : "TCP");
        

        ao2_ref(ser, -1);
        ser = NULL;

        return NULL;
}


/*!
 * 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, char *tag)
{
        ao2_t_ref(peer, -1, tag);
        return NULL;
}

static struct sip_peer *ref_peer(struct sip_peer *peer, char *tag)
{
        ao2_t_ref(peer, 1, tag);
        return peer;
}

/*! \brief maintain proper refcounts for a sip_pvt's outboundproxy
 *
 * This function sets pvt's outboundproxy pointer to the one referenced
 * by the proxy parameter. Because proxy may be a refcounted object, and
 * because pvt's old outboundproxy may also be a refcounted object, we need
 * to maintain the proper refcounts.
 *
 * \param pvt The sip_pvt for which we wish to set the outboundproxy
 * \param proxy The sip_proxy which we will point pvt towards.
 * \return Returns void
 */
static void ref_proxy(struct sip_pvt *pvt, struct sip_proxy *proxy)
{
        struct sip_proxy *old_obproxy = pvt->outboundproxy;
        /* The global_outboundproxy is statically allocated, and so
         * we don't ever need to adjust refcounts for it
         */
        if (proxy && proxy != &global_outboundproxy) {
                ao2_ref(proxy, +1);
        }
        pvt->outboundproxy = proxy;
        if (old_obproxy && old_obproxy != &global_outboundproxy) {
                ao2_ref(old_obproxy, -1);
        }
}

/*!
 * \brief Unlink a dialog from the dialogs container, as well as any other places
 * that it may be currently stored.
 *
 * \note A reference to the dialog must be held before calling this function, and this
 * function does not release that reference.
 */
static void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist)
{
        struct sip_pkt *cp;

        dialog_ref(dialog, "Let's bump the count in the unlink so it doesn't accidentally become dead before we are done");

        ao2_t_unlink(dialogs, dialog, "unlinking dialog via ao2_unlink");

        /* Unlink us from the owner (channel) if we have one */
        if (dialog->owner) {
                if (lockowner)
                        ast_channel_lock(dialog->owner);
                ast_debug(1, "Detaching from channel %s\n", dialog->owner->name);
                dialog->owner->tech_pvt = dialog_unref(dialog->owner->tech_pvt, "resetting channel dialog ptr in unlink_all");
                if (lockowner)
                        ast_channel_unlock(dialog->owner);
        }
        if (dialog->registry) {
                if (dialog->registry->call == dialog)
                        dialog->registry->call = dialog_unref(dialog->registry->call, "nulling out the registry's call dialog field in unlink_all");
                dialog->registry = registry_unref(dialog->registry, "delete dialog->registry");
        }
        if (dialog->stateid > -1) {
                ast_extension_state_del(dialog->stateid, NULL);
                dialog_unref(dialog, "removing extension_state, should unref the associated dialog ptr that was stored there.");
                dialog->stateid = -1; /* shouldn't we 'zero' this out? */
        }
        /* Remove link from peer to subscription of MWI */
        if (dialog->relatedpeer && dialog->relatedpeer->mwipvt == dialog)
                dialog->relatedpeer->mwipvt = dialog_unref(dialog->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
        if (dialog->relatedpeer && dialog->relatedpeer->call == dialog)
                dialog->relatedpeer->call = dialog_unref(dialog->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");

        /* remove all current packets in this dialog */
        while((cp = dialog->packets)) {
                dialog->packets = dialog->packets->next;
                AST_SCHED_DEL(sched, cp->retransid);
                dialog_unref(cp->owner, "remove all current packets in this dialog, and the pointer to the dialog too as part of __sip_destroy");
                ast_free(cp);
        }

        AST_SCHED_DEL_UNREF(sched, dialog->waitid, dialog_unref(dialog, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));

        AST_SCHED_DEL_UNREF(sched, dialog->initid, dialog_unref(dialog, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));
        
        if (dialog->autokillid > -1)
                AST_SCHED_DEL_UNREF(sched, dialog->autokillid, dialog_unref(dialog, "when you delete the autokillid sched, you should dec the refcount for the stored dialog ptr"));

        dialog_unref(dialog, "Let's unbump the count in the unlink so the poor pvt can disappear if it is time");
        return NULL;
}

static void *registry_unref(struct sip_registry *reg, char *tag)
{
        ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount - 1);
        ASTOBJ_UNREF(reg, sip_registry_destroy);
        return NULL;
}

/*! \brief Add object reference to SIP registry */
static struct sip_registry *registry_addref(struct sip_registry *reg, char *tag)
{
        ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount + 1);
        return ASTOBJ_REF(reg); /* Add pointer to registry in packet */
}

/*! \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,
};

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


/*! \brief Convert transfer status to string */
static const char *referstatus2str(enum referstatus rstatus)
{
        return map_x_s(referstatusstrings, rstatus, "");
}

static inline void pvt_set_needdestroy(struct sip_pvt *pvt, const char *reason)
{
        append_history(pvt, "NeedDestroy", "Setting needdestroy because %s", reason);
        pvt->needdestroy = 1;
}

/*! \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 (p->initreq.headers)
                ast_debug(1, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
        else
                ast_debug(1, "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 (req->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)
{
        ast_debug(3, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
        dialog->alreadygone = 1;
}

/*! Resolve DNS srv name or host name in a sip_proxy structure */
static int proxy_update(struct sip_proxy *proxy)
{
        /* if it's actually an IP address and not a name,
           there's no need for a managed lookup */
        if (!inet_aton(proxy->name, &proxy->ip.sin_addr)) {
                /* Ok, not an IP address, then let's check if it's a domain or host */
                /* XXX Todo - if we have proxy port, don't do SRV */
                if (ast_get_ip_or_srv(&proxy->ip, proxy->name, sip_cfg.srvlookup ? "_sip._udp" : NULL) < 0) {
                        ast_log(LOG_WARNING, "Unable to locate host '%s'\n", proxy->name);
                        return FALSE;
                }
        }
        proxy->last_dnsupdate = time(NULL);
        return TRUE;
}

/*! \brief Allocate and initialize sip proxy */
static struct sip_proxy *proxy_allocate(char *name, char *port, int force)
{
        struct sip_proxy *proxy;
        proxy = ao2_alloc(sizeof(*proxy), NULL);
      &n