The large GULP->PJSIP renaming effort.

The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:

* The word "Gulp" in dialstrings, functions, and CLI commands is now
  "PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 0 insertions, 1502 deletions

diff --git a/include/asterisk/res_sip.h b/include/asterisk/res_sip.h
deleted file mode 100644
index 23d1a641e..000000000
--- a/include/asterisk/res_sip.h
+++ /dev/null
@@ -1,1502 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2013, Digium, Inc.
- *
- * Mark Michelson <mmichelson@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.
- */
-
-#ifndef _RES_SIP_H
-#define _RES_SIP_H
-
-#include "asterisk/stringfields.h"
-/* Needed for struct ast_sockaddr */
-#include "asterisk/netsock2.h"
-/* Needed for linked list macros */
-#include "asterisk/linkedlists.h"
-/* Needed for ast_party_id */
-#include "asterisk/channel.h"
-/* Needed for ast_sorcery */
-#include "asterisk/sorcery.h"
-/* Needed for ast_dnsmgr */
-#include "asterisk/dnsmgr.h"
-/* Needed for ast_endpoint */
-#include "asterisk/endpoints.h"
-/* Needed for ast_t38_ec_modes */
-#include "asterisk/udptl.h"
-/* Needed for pj_sockaddr */
-#include <pjlib.h>
-/* Needed for ast_rtp_dtls_cfg struct */
-#include "asterisk/rtp_engine.h"
-
-/* Forward declarations of PJSIP stuff */
-struct pjsip_rx_data;
-struct pjsip_module;
-struct pjsip_tx_data;
-struct pjsip_dialog;
-struct pjsip_transport;
-struct pjsip_tpfactory;
-struct pjsip_tls_setting;
-struct pjsip_tpselector;
-
-/*!
- * \brief Structure for SIP transport information
- */
-struct ast_sip_transport_state {
-	/*! \brief Transport itself */
-	struct pjsip_transport *transport;
-
-	/*! \brief Transport factory */
-	struct pjsip_tpfactory *factory;
-};
-
-#define SIP_SORCERY_DOMAIN_ALIAS_TYPE "domain_alias"
-
-/*!
- * Details about a SIP domain alias
- */
-struct ast_sip_domain_alias {
-	/*! Sorcery object details */
-	SORCERY_OBJECT(details);
-	AST_DECLARE_STRING_FIELDS(
-		/*! Domain to be aliased to */
-		AST_STRING_FIELD(domain);
-	);
-};
-
-/*! \brief Maximum number of ciphers supported for a TLS transport */
-#define SIP_TLS_MAX_CIPHERS 64
-
-/*
- * \brief Transport to bind to
- */
-struct ast_sip_transport {
-	/*! Sorcery object details */
-	SORCERY_OBJECT(details);
-	AST_DECLARE_STRING_FIELDS(
-		/*! Certificate of authority list file */
-		AST_STRING_FIELD(ca_list_file);
-		/*! Public certificate file */
-		AST_STRING_FIELD(cert_file);
-		/*! Optional private key of the certificate file */
-		AST_STRING_FIELD(privkey_file);
-		/*! Password to open the private key */
-		AST_STRING_FIELD(password);
-		/*! External signaling address */
-		AST_STRING_FIELD(external_signaling_address);
-		/*! External media address */
-		AST_STRING_FIELD(external_media_address);
-		/*! Optional domain to use for messages if provided could not be found */
-		AST_STRING_FIELD(domain);
-		);
-	/*! Type of transport */
-	enum ast_transport type;
-	/*! Address and port to bind to */
-	pj_sockaddr host;
-	/*! Number of simultaneous asynchronous operations */
-	unsigned int async_operations;
-	/*! Optional external port for signaling */
-	unsigned int external_signaling_port;
-	/*! TLS settings */
-	pjsip_tls_setting tls;
-	/*! Configured TLS ciphers */
-	pj_ssl_cipher ciphers[SIP_TLS_MAX_CIPHERS];
-	/*! Optional local network information, used for NAT purposes */
-	struct ast_ha *localnet;
-	/*! DNS manager for refreshing the external address */
-	struct ast_dnsmgr_entry *external_address_refresher;
-	/*! Optional external address information */
-	struct ast_sockaddr external_address;
-	/*! Transport state information */
-	struct ast_sip_transport_state *state;
-	/*! QOS DSCP TOS bits */
-	unsigned int tos;
-	/*! QOS COS value */
-	unsigned int cos;
-};
-
-/*!
- * \brief Structure for SIP nat hook information
- */
-struct ast_sip_nat_hook {
-	/*! Sorcery object details */
-	SORCERY_OBJECT(details);
-	/*! Callback for when a message is going outside of our local network */
-	void (*outgoing_external_message)(struct pjsip_tx_data *tdata, struct ast_sip_transport *transport);
-};
-
-/*!
- * \brief Contact associated with an address of record
- */
-struct ast_sip_contact {
-	/*! Sorcery object details, the id is the aor name plus a random string */
-	SORCERY_OBJECT(details);
-	AST_DECLARE_STRING_FIELDS(
-		/*! Full URI of the contact */
-		AST_STRING_FIELD(uri);
-	);
-	/*! Absolute time that this contact is no longer valid after */
-	struct timeval expiration_time;
-	/*! Frequency to send OPTIONS requests to contact. 0 is disabled. */
-	unsigned int qualify_frequency;
-	/*! If true authenticate the qualify if needed */
-	int authenticate_qualify;
-};
-
-#define CONTACT_STATUS "contact_status"
-
-/*!
- * \brief Status type for a contact.
- */
-enum ast_sip_contact_status_type {
-	UNAVAILABLE,
-	AVAILABLE
-};
-
-/*!
- * \brief A contact's status.
- *
- * \detail Maintains a contact's current status and round trip time
- *         if available.
- */
-struct ast_sip_contact_status {
-	SORCERY_OBJECT(details);
-	/*! Current status for a contact (default - unavailable) */
-	enum ast_sip_contact_status_type status;
-	/*! The round trip start time set before sending a qualify request */
-	struct timeval rtt_start;
-	/*! The round trip time in microseconds */
-	int64_t rtt;
-};
-
-/*!
- * \brief A transport to be used for messages to a contact
- */
-struct ast_sip_contact_transport {
-	AST_DECLARE_STRING_FIELDS(
-		/*! Full URI of the contact */
-		AST_STRING_FIELD(uri);
-	);
-	pjsip_transport *transport;
-};
-
-/*!
- * \brief A SIP address of record
- */
-struct ast_sip_aor {
-	/*! Sorcery object details, the id is the AOR name */
-	SORCERY_OBJECT(details);
-	AST_DECLARE_STRING_FIELDS(
-		/*! Voicemail boxes for this AOR */
-		AST_STRING_FIELD(mailboxes);
-	);
-	/*! Minimum expiration time */
-	unsigned int minimum_expiration;
-	/*! Maximum expiration time */
-	unsigned int maximum_expiration;
-	/*! Default contact expiration if one is not provided in the contact */
-	unsigned int default_expiration;
-	/*! Frequency to send OPTIONS requests to AOR contacts. 0 is disabled. */
-	unsigned int qualify_frequency;
-	/*! If true authenticate the qualify if needed */
-	int authenticate_qualify;
-	/*! Maximum number of external contacts, 0 to disable */
-	unsigned int max_contacts;
-	/*! Whether to remove any existing contacts not related to an incoming REGISTER when it comes in */
-	unsigned int remove_existing;
-	/*! Any permanent configured contacts */
-	struct ao2_container *permanent_contacts;
-};
-
-/*!
- * \brief DTMF modes for SIP endpoints
- */
-enum ast_sip_dtmf_mode {
-	/*! No DTMF to be used */
-	AST_SIP_DTMF_NONE,
-	/* XXX Should this be 2833 instead? */
-	/*! Use RFC 4733 events for DTMF */
-	AST_SIP_DTMF_RFC_4733,
-	/*! Use DTMF in the audio stream */
-	AST_SIP_DTMF_INBAND,
-	/*! Use SIP INFO DTMF (blech) */
-	AST_SIP_DTMF_INFO,
-};
-
-/*!
- * \brief Methods of storing SIP digest authentication credentials.
- *
- * Note that both methods result in MD5 digest authentication being
- * used. The two methods simply alter how Asterisk determines the
- * credentials for a SIP authentication
- */
-enum ast_sip_auth_type {
-	/*! Credentials stored as a username and password combination */
-	AST_SIP_AUTH_TYPE_USER_PASS,
-	/*! Credentials stored as an MD5 sum */
-	AST_SIP_AUTH_TYPE_MD5,
-	/*! Credentials not stored this is a fake auth */
-	AST_SIP_AUTH_TYPE_ARTIFICIAL
-};
-
-#define SIP_SORCERY_AUTH_TYPE "auth"
-
-struct ast_sip_auth {
-	/* Sorcery ID of the auth is its name */
-	SORCERY_OBJECT(details);
-	AST_DECLARE_STRING_FIELDS(
-		/* Identification for these credentials */
-		AST_STRING_FIELD(realm);
-		/* Authentication username */
-		AST_STRING_FIELD(auth_user);
-		/* Authentication password */
-		AST_STRING_FIELD(auth_pass);
-		/* Authentication credentials in MD5 format (hash of user:realm:pass) */
-		AST_STRING_FIELD(md5_creds);
-	);
-	/* The time period (in seconds) that a nonce may be reused */
-	unsigned int nonce_lifetime;
-	/* Used to determine what to use when authenticating */
-	enum ast_sip_auth_type type;
-};
-
-struct ast_sip_auth_array {
-	/*! Array of Sorcery IDs of auth sections */
-	const char **names;
-	/*! Number of credentials in the array */
-	unsigned int num;
-};
-
-/*!
- * \brief Different methods by which incoming requests can be matched to endpoints
- */
-enum ast_sip_endpoint_identifier_type {
-	/*! Identify based on user name in From header */
-	AST_SIP_ENDPOINT_IDENTIFY_BY_USERNAME = (1 << 0),
-};
-
-enum ast_sip_session_refresh_method {
-	/*! Use reinvite to negotiate direct media */
-	AST_SIP_SESSION_REFRESH_METHOD_INVITE,
-	/*! Use UPDATE to negotiate direct media */
-	AST_SIP_SESSION_REFRESH_METHOD_UPDATE,
-};
-
-enum ast_sip_direct_media_glare_mitigation {
-	/*! Take no special action to mitigate reinvite glare */
-	AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE,
-	/*! Do not send an initial direct media session refresh on outgoing call legs
-	 * Subsequent session refreshes will be sent no matter the session direction
-	 */
-	AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING,
-	/*! Do not send an initial direct media session refresh on incoming call legs
-	 * Subsequent session refreshes will be sent no matter the session direction
-	 */
-	AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING,
-};
-
-enum ast_sip_session_media_encryption {
-	/*! Invalid media encryption configuration */
-	AST_SIP_MEDIA_TRANSPORT_INVALID = 0,
-	/*! Do not allow any encryption of session media */
-	AST_SIP_MEDIA_ENCRYPT_NONE,
-	/*! Offer SDES-encrypted session media */
-	AST_SIP_MEDIA_ENCRYPT_SDES,
-	/*! Offer encrypted session media with datagram TLS key exchange */
-	AST_SIP_MEDIA_ENCRYPT_DTLS,
-};
-
-/*!
- * \brief Session timers options
- */
-struct ast_sip_timer_options {
-	/*! Minimum session expiration period, in seconds */
-	unsigned int min_se;
-	/*! Session expiration period, in seconds */
-	unsigned int sess_expires;
-};
-
-/*!
- * \brief Endpoint configuration for SIP extensions.
- *
- * SIP extensions, in this case refers to features
- * indicated in Supported or Required headers.
- */
-struct ast_sip_endpoint_extensions {
-	/*! Enabled SIP extensions */
-	unsigned int flags;
-	/*! Timer options */
-	struct ast_sip_timer_options timer;
-};
-
-/*!
- * \brief Endpoint configuration for unsolicited MWI
- */
-struct ast_sip_mwi_configuration {
-	AST_DECLARE_STRING_FIELDS(
-		/*! Configured voicemail boxes for this endpoint. Used for MWI */
-		AST_STRING_FIELD(mailboxes);
-		/*! Username to use when sending MWI NOTIFYs to this endpoint */
-		AST_STRING_FIELD(fromuser);
-	);
-	/* Should mailbox states be combined into a single notification? */
-	unsigned int aggregate;
-};
-
-/*!
- * \brief Endpoint subscription configuration
- */
-struct ast_sip_endpoint_subscription_configuration {
-	/*! Indicates if endpoint is allowed to initiate subscriptions */
-	unsigned int allow;
-	/*! The minimum allowed expiration for subscriptions from endpoint */
-	unsigned int minexpiry;
-	/*! Message waiting configuration */
-	struct ast_sip_mwi_configuration mwi;
-};
-
-/*!
- * \brief NAT configuration options for endpoints
- */
-struct ast_sip_endpoint_nat_configuration {
-	/*! Whether to force using the source IP address/port for sending responses */
-	unsigned int force_rport;
-	/*! Whether to rewrite the Contact header with the source IP address/port or not */
-	unsigned int rewrite_contact;
-};
-
-/*!
- * \brief Party identification options for endpoints
- *
- * This includes caller ID, connected line, and redirecting-related options
- */
-struct ast_sip_endpoint_id_configuration {
-	struct ast_party_id self;
-	/*! Do we accept identification information from this endpoint */
-	unsigned int trust_inbound;
-	/*! Do we send private identification information to this endpoint? */
-	unsigned int trust_outbound;
-	/*! Do we send P-Asserted-Identity headers to this endpoint? */
-	unsigned int send_pai;
-	/*! Do we send Remote-Party-ID headers to this endpoint? */
-	unsigned int send_rpid;
-	/*! Do we add Diversion headers to applicable outgoing requests/responses? */
-	unsigned int send_diversion;
-	/*! When performing connected line update, which method should be used */
-	enum ast_sip_session_refresh_method refresh_method;
-};
-
-/*!
- * \brief Call pickup configuration options for endpoints
- */
-struct ast_sip_endpoint_pickup_configuration {
-	/*! Call group */
-	ast_group_t callgroup;
-	/*! Pickup group */
-	ast_group_t pickupgroup;
-	/*! Named call group */
-	struct ast_namedgroups *named_callgroups;
-	/*! Named pickup group */
-	struct ast_namedgroups *named_pickupgroups;
-};
-
-/*!
- * \brief Configuration for one-touch INFO recording
- */
-struct ast_sip_info_recording_configuration {
-	AST_DECLARE_STRING_FIELDS(
-		/*! Feature to enact when one-touch recording INFO with Record: On is received */
-		AST_STRING_FIELD(onfeature);
-		/*! Feature to enact when one-touch recording INFO with Record: Off is received */
-		AST_STRING_FIELD(offfeature);
-	);
-	/*! Is one-touch recording permitted? */
-	unsigned int enabled;
-};
-
-/*!
- * \brief Endpoint configuration options for INFO packages
- */
-struct ast_sip_endpoint_info_configuration {
-	/*! Configuration for one-touch recording */
-	struct ast_sip_info_recording_configuration recording;
-};
-
-/*!
- * \brief RTP configuration for SIP endpoints
- */
-struct ast_sip_media_rtp_configuration {
-	AST_DECLARE_STRING_FIELDS(
-		/*! Configured RTP engine for this endpoint. */
-		AST_STRING_FIELD(engine);
-	);
-	/*! Whether IPv6 RTP is enabled or not */
-	unsigned int ipv6;
-	/*! Whether symmetric RTP is enabled or not */
-	unsigned int symmetric;
-	/*! Whether ICE support is enabled or not */
-	unsigned int ice_support;
-	/*! Whether to use the "ptime" attribute received from the endpoint or not */
-	unsigned int use_ptime;
-	/*! Do we use AVPF exclusively for this endpoint? */
-	unsigned int use_avpf;
-	/*! \brief DTLS-SRTP configuration information */
-	struct ast_rtp_dtls_cfg dtls_cfg;
-	/*! Should SRTP use a 32 byte tag instead of an 80 byte tag? */
-	unsigned int srtp_tag_32;
-	/*! Do we use media encryption? what type? */
-	enum ast_sip_session_media_encryption encryption;
-};
-
-/*!
- * \brief Direct media options for SIP endpoints
- */
-struct ast_sip_direct_media_configuration {
-	/*! Boolean indicating if direct_media is permissible */
-	unsigned int enabled;
-	/*! When using direct media, which method should be used */
-	enum ast_sip_session_refresh_method method;
-	/*! Take steps to mitigate glare for direct media */
-	enum ast_sip_direct_media_glare_mitigation glare_mitigation;
-	/*! Do not attempt direct media session refreshes if a media NAT is detected */
-	unsigned int disable_on_nat;
-};
-
-struct ast_sip_t38_configuration {
-	/*! Whether T.38 UDPTL support is enabled or not */
-	unsigned int enabled;
-	/*! Error correction setting for T.38 UDPTL */
-	enum ast_t38_ec_modes error_correction;
-	/*! Explicit T.38 max datagram value, may be 0 to indicate the remote side can be trusted */
-	unsigned int maxdatagram;
-	/*! Whether NAT Support is enabled for T.38 UDPTL sessions or not */
-	unsigned int nat;
-	/*! Whether to use IPv6 for UDPTL or not */
-	unsigned int ipv6;
-};
-
-/*!
- * \brief Media configuration for SIP endpoints
- */
-struct ast_sip_endpoint_media_configuration {
-	AST_DECLARE_STRING_FIELDS(
-		/*! Optional external media address to use in SDP */
-		AST_STRING_FIELD(external_address);
-		/*! SDP origin username */
-		AST_STRING_FIELD(sdpowner);
-		/*! SDP session name */
-		AST_STRING_FIELD(sdpsession);
-	);
-	/*! RTP media configuration */
-	struct ast_sip_media_rtp_configuration rtp;
-	/*! Direct media options */
-	struct ast_sip_direct_media_configuration direct_media;
-	/*! T.38 (FoIP) options */
-	struct ast_sip_t38_configuration t38;
-	/*! Codec preferences */
-	struct ast_codec_pref prefs;
-	/*! Configured codecs */
-	struct ast_format_cap *codecs;
-	/*! DSCP TOS bits for audio streams */
-	unsigned int tos_audio;
-	/*! Priority for audio streams */
-	unsigned int cos_audio;
-	/*! DSCP TOS bits for video streams */
-	unsigned int tos_video;
-	/*! Priority for video streams */
-	unsigned int cos_video;
-};
-
-/*!
- * \brief An entity with which Asterisk communicates
- */
-struct ast_sip_endpoint {
-	SORCERY_OBJECT(details);
-	AST_DECLARE_STRING_FIELDS(
-		/*! Context to send incoming calls to */
-		AST_STRING_FIELD(context);
-		/*! Name of an explicit transport to use */
-		AST_STRING_FIELD(transport);
-		/*! Outbound proxy to use */
-		AST_STRING_FIELD(outbound_proxy);
-		/*! Explicit AORs to dial if none are specified */
-		AST_STRING_FIELD(aors);
-		/*! Musiconhold class to suggest that the other side use when placing on hold */
-		AST_STRING_FIELD(mohsuggest);
-		/*! Configured tone zone for this endpoint. */
-		AST_STRING_FIELD(zone);
-		/*! Configured language for this endpoint. */
-		AST_STRING_FIELD(language);
-		/*! Default username to place in From header */
-		AST_STRING_FIELD(fromuser);
-		/*! Domain to place in From header */
-		AST_STRING_FIELD(fromdomain);
-	);
-	/*! Configuration for extensions */
-	struct ast_sip_endpoint_extensions extensions;
-	/*! Configuration relating to media */
-	struct ast_sip_endpoint_media_configuration media;
-	/*! SUBSCRIBE/NOTIFY configuration options */
-	struct ast_sip_endpoint_subscription_configuration subscription;
-	/*! NAT configuration */
-	struct ast_sip_endpoint_nat_configuration nat;
-	/*! Party identification options */
-	struct ast_sip_endpoint_id_configuration id;
-	/*! Configuration options for INFO packages */
-	struct ast_sip_endpoint_info_configuration info;
-	/*! Call pickup configuration */
-	struct ast_sip_endpoint_pickup_configuration pickup;
-	/*! Inbound authentication credentials */
-	struct ast_sip_auth_array inbound_auths;
-	/*! Outbound authentication credentials */
-	struct ast_sip_auth_array outbound_auths;
-	/*! DTMF mode to use with this endpoint */
-	enum ast_sip_dtmf_mode dtmf;
-	/*! Method(s) by which the endpoint should be identified. */
-	enum ast_sip_endpoint_identifier_type ident_method;
-	/*! Boolean indicating if ringing should be sent as inband progress */
-	unsigned int inband_progress;
-	/*! Pointer to the persistent Asterisk endpoint */
-	struct ast_endpoint *persistent;
-	/*! The number of channels at which busy device state is returned */
-	unsigned int devicestate_busy_at;
-	/*! Whether fax detection is enabled or not (CNG tone detection) */
-	unsigned int faxdetect;
-	/*! Determines if transfers (using REFER) are allowed by this endpoint */
-	unsigned int allowtransfer;
-};
-
-/*!
- * \brief Initialize an auth array with the configured values.
- *
- * \param array Array to initialize
- * \param auth_names Comma-separated list of names to set in the array
- * \retval 0 Success
- * \retval non-zero Failure
- */
-int ast_sip_auth_array_init(struct ast_sip_auth_array *array, const char *auth_names);
-
-/*!
- * \brief Free contents of an auth array.
- *
- * \param array Array whose contents are to be freed
- */
-void ast_sip_auth_array_destroy(struct ast_sip_auth_array *array);
-
-/*!
- * \brief Possible returns from ast_sip_check_authentication
- */
-enum ast_sip_check_auth_result {
-    /*! Authentication needs to be challenged */
-    AST_SIP_AUTHENTICATION_CHALLENGE,
-    /*! Authentication succeeded */
-    AST_SIP_AUTHENTICATION_SUCCESS,
-    /*! Authentication failed */
-    AST_SIP_AUTHENTICATION_FAILED,
-    /*! Authentication encountered some internal error */
-    AST_SIP_AUTHENTICATION_ERROR,
-};
-
-/*!
- * \brief An interchangeable way of handling digest authentication for SIP.
- *
- * An authenticator is responsible for filling in the callbacks provided below. Each is called from a publicly available
- * function in res_sip. The authenticator can use configuration or other local policy to determine whether authentication
- * should take place and what credentials should be used when challenging and authenticating a request.
- */
-struct ast_sip_authenticator {
-    /*!
-     * \brief Check if a request requires authentication
-     * See ast_sip_requires_authentication for more details
-     */
-    int (*requires_authentication)(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
-	/*!
-	 * \brief Check that an incoming request passes authentication.
-	 *
-	 * The tdata parameter is useful for adding information such as digest challenges.
-	 *
-	 * \param endpoint The endpoint sending the incoming request
-	 * \param rdata The incoming request
-	 * \param tdata Tentative outgoing request.
-	 */
-	enum ast_sip_check_auth_result (*check_authentication)(struct ast_sip_endpoint *endpoint,
-			pjsip_rx_data *rdata, pjsip_tx_data *tdata);
-};
-
-/*!
- * \brief an interchangeable way of responding to authentication challenges
- *
- * An outbound authenticator takes incoming challenges and formulates a new SIP request with
- * credentials.
- */
-struct ast_sip_outbound_authenticator {
-	/*!
-	 * \brief Create a new request with authentication credentials
-	 *
-	 * \param auths An array of IDs of auth sorcery objects
-	 * \param challenge The SIP response with authentication challenge(s)
-	 * \param tsx The transaction in which the challenge was received
-	 * \param new_request The new SIP request with challenge response(s)
-	 * \retval 0 Successfully created new request
-	 * \retval -1 Failed to create a new request
-	 */
-	int (*create_request_with_auth)(const struct ast_sip_auth_array *auths, struct pjsip_rx_data *challenge,
-			struct pjsip_transaction *tsx, struct pjsip_tx_data **new_request);
-};
-
-/*!
- * \brief An entity responsible for identifying the source of a SIP message
- */
-struct ast_sip_endpoint_identifier {
-    /*!
-     * \brief Callback used to identify the source of a message.
-     * See ast_sip_identify_endpoint for more details
-     */
-    struct ast_sip_endpoint *(*identify_endpoint)(pjsip_rx_data *rdata);
-};
-
-#define SIP_SORCERY_SECURITY_TYPE "security"
-
-/*!
- * \brief SIP security details and configuration.
- */
-struct ast_sip_security {
-	SORCERY_OBJECT(details);
-	struct ast_acl_list *acl;
-	struct ast_acl_list *contact_acl;
-};
-
-/*!
- * \brief Register a SIP service in Asterisk.
- *
- * This is more-or-less a wrapper around pjsip_endpt_register_module().
- * Registering a service makes it so that PJSIP will call into the
- * service at appropriate times. For more information about PJSIP module
- * callbacks, see the PJSIP documentation. Asterisk modules that call
- * this function will likely do so at module load time.
- *
- * \param module The module that is to be registered with PJSIP
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_register_service(pjsip_module *module);
-
-/*!
- * This is the opposite of ast_sip_register_service().  Unregistering a
- * service means that PJSIP will no longer call into the module any more.
- * This will likely occur when an Asterisk module is unloaded.
- *
- * \param module The PJSIP module to unregister
- */
-void ast_sip_unregister_service(pjsip_module *module);
-
-/*!
- * \brief Register a SIP authenticator
- *
- * An authenticator has three main purposes:
- * 1) Determining if authentication should be performed on an incoming request
- * 2) Gathering credentials necessary for issuing an authentication challenge
- * 3) Authenticating a request that has credentials
- *
- * Asterisk provides a default authenticator, but it may be replaced by a
- * custom one if desired.
- *
- * \param auth The authenticator to register
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_register_authenticator(struct ast_sip_authenticator *auth);
-
-/*!
- * \brief Unregister a SIP authenticator
- *
- * When there is no authenticator registered, requests cannot be challenged
- * or authenticated.
- *
- * \param auth The authenticator to unregister
- */
-void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth);
-
- /*!
- * \brief Register an outbound SIP authenticator
- *
- * An outbound authenticator is responsible for creating responses to
- * authentication challenges by remote endpoints.
- *
- * \param auth The authenticator to register
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *outbound_auth);
-
-/*!
- * \brief Unregister an outbound SIP authenticator
- *
- * When there is no outbound authenticator registered, authentication challenges
- * will be handled as any other final response would be.
- *
- * \param auth The authenticator to unregister
- */
-void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth);
-
-/*!
- * \brief Register a SIP endpoint identifier
- *
- * An endpoint identifier's purpose is to determine which endpoint a given SIP
- * message has come from.
- *
- * Multiple endpoint identifiers may be registered so that if an endpoint
- * cannot be identified by one identifier, it may be identified by another.
- *
- * Asterisk provides two endpoint identifiers. One identifies endpoints based
- * on the user part of the From header URI. The other identifies endpoints based
- * on the source IP address.
- *
- * If the order in which endpoint identifiers is run is important to you, then
- * be sure to load individual endpoint identifier modules in the order you wish
- * for them to be run in modules.conf
- *
- * \param identifier The SIP endpoint identifier to register
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier);
-
-/*!
- * \brief Unregister a SIP endpoint identifier
- *
- * This stops an endpoint identifier from being used.
- *
- * \param identifier The SIP endoint identifier to unregister
- */
-void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier);
-
-/*!
- * \brief Allocate a new SIP endpoint
- *
- * This will return an endpoint with its refcount increased by one. This reference
- * can be released using ao2_ref().
- *
- * \param name The name of the endpoint.
- * \retval NULL Endpoint allocation failed
- * \retval non-NULL The newly allocated endpoint
- */
-void *ast_sip_endpoint_alloc(const char *name);
-
-/*!
- * \brief Get a pointer to the PJSIP endpoint.
- *
- * This is useful when modules have specific information they need
- * to register with the PJSIP core.
- * \retval NULL endpoint has not been created yet.
- * \retval non-NULL PJSIP endpoint.
- */
-pjsip_endpoint *ast_sip_get_pjsip_endpoint(void);
-
-/*!
- * \brief Get a pointer to the SIP sorcery structure.
- *
- * \retval NULL sorcery has not been initialized
- * \retval non-NULL sorcery structure
- */
-struct ast_sorcery *ast_sip_get_sorcery(void);
-
-/*!
- * \brief Initialize transport support on a sorcery instance
- *
- * \param sorcery The sorcery instance
- *
- * \retval -1 failure
- * \retval 0 success
- */
-int ast_sip_initialize_sorcery_transport(struct ast_sorcery *sorcery);
-
-/*!
- * \brief Initialize qualify support on a sorcery instance
- *
- * \param sorcery The sorcery instance
- *
- * \retval -1 failure
- * \retval 0 success
- */
-int ast_sip_initialize_sorcery_qualify(struct ast_sorcery *sorcery);
-
-/*!
- * \brief Initialize location support on a sorcery instance
- *
- * \param sorcery The sorcery instance
- *
- * \retval -1 failure
- * \retval 0 success
- */
-int ast_sip_initialize_sorcery_location(struct ast_sorcery *sorcery);
-
-/*!
- * \brief Retrieve a named AOR
- *
- * \param aor_name Name of the AOR
- *
- * \retval NULL if not found
- * \retval non-NULL if found
- */
-struct ast_sip_aor *ast_sip_location_retrieve_aor(const char *aor_name);
-
-/*!
- * \brief Retrieve the first bound contact for an AOR
- *
- * \param aor Pointer to the AOR
- * \retval NULL if no contacts available
- * \retval non-NULL if contacts available
- */
-struct ast_sip_contact *ast_sip_location_retrieve_first_aor_contact(const struct ast_sip_aor *aor);
-
-/*!
- * \brief Retrieve all contacts currently available for an AOR
- *
- * \param aor Pointer to the AOR
- *
- * \retval NULL if no contacts available
- * \retval non-NULL if contacts available
- */
-struct ao2_container *ast_sip_location_retrieve_aor_contacts(const struct ast_sip_aor *aor);
-
-/*!
- * \brief Retrieve the first bound contact from a list of AORs
- *
- * \param aor_list A comma-separated list of AOR names
- * \retval NULL if no contacts available
- * \retval non-NULL if contacts available
- */
-struct ast_sip_contact *ast_sip_location_retrieve_contact_from_aor_list(const char *aor_list);
-
-/*!
- * \brief Retrieve a named contact
- *
- * \param contact_name Name of the contact
- *
- * \retval NULL if not found
- * \retval non-NULL if found
- */
-struct ast_sip_contact *ast_sip_location_retrieve_contact(const char *contact_name);
-
-/*!
- * \brief Add a transport for a contact to use
- */
-
-void ast_sip_location_add_contact_transport(struct ast_sip_contact_transport *ct);
-
-/*!
- * \brief Delete a transport for a contact that went away
- */
-void ast_sip_location_delete_contact_transport(struct ast_sip_contact_transport *ct);
-
-/*!
- * \brief Retrieve a contact_transport, by URI
- *
- * \param contact_uri URI of the contact
- *
- * \retval NULL if not found
- * \retval non-NULL if found
- */
-struct ast_sip_contact_transport *ast_sip_location_retrieve_contact_transport_by_uri(const char *contact_uri);
-
-/*!
- * \brief Retrieve a contact_transport, by transport
- *
- * \param transport transport the contact uses
- *
- * \retval NULL if not found
- * \retval non-NULL if found
- */
-struct ast_sip_contact_transport *ast_sip_location_retrieve_contact_transport_by_transport(pjsip_transport *transport);
-
-/*!
- * \brief Add a new contact to an AOR
- *
- * \param aor Pointer to the AOR
- * \param uri Full contact URI
- * \param expiration_time Optional expiration time of the contact
- *
- * \retval -1 failure
- * \retval 0 success
- */
-int ast_sip_location_add_contact(struct ast_sip_aor *aor, const char *uri, struct timeval expiration_time);
-
-/*!
- * \brief Update a contact
- *
- * \param contact New contact object with details
- *
- * \retval -1 failure
- * \retval 0 success
- */
-int ast_sip_location_update_contact(struct ast_sip_contact *contact);
-
-/*!
-* \brief Delete a contact
-*
-* \param contact Contact object to delete
-*
-* \retval -1 failure
-* \retval 0 success
-*/
-int ast_sip_location_delete_contact(struct ast_sip_contact *contact);
-
-/*!
- * \brief Initialize domain aliases support on a sorcery instance
- *
- * \param sorcery The sorcery instance
- *
- * \retval -1 failure
- * \retval 0 success
- */
-int ast_sip_initialize_sorcery_domain_alias(struct ast_sorcery *sorcery);
-
-/*!
- * \brief Initialize authentication support on a sorcery instance
- *
- * \param sorcery The sorcery instance
- *
- * \retval -1 failure
- * \retval 0 success
- */
-int ast_sip_initialize_sorcery_auth(struct ast_sorcery *sorcery);
-
-/*!
- * \brief Initialize security support on a sorcery instance
- *
- * \param sorcery The sorcery instance
- *
- * \retval -1 failure
- * \retval 0 success
- */
-int ast_sip_initialize_sorcery_security(struct ast_sorcery *sorcery);
-
-/*!
- * \brief Callback called when an outbound request with authentication credentials is to be sent in dialog
- *
- * This callback will have the created request on it. The callback's purpose is to do any extra
- * housekeeping that needs to be done as well as to send the request out.
- *
- * This callback is only necessary if working with a PJSIP API that sits between the application
- * and the dialog layer.
- *
- * \param dlg The dialog to which the request belongs
- * \param tdata The created request to be sent out
- * \param user_data Data supplied with the callback
- *
- * \retval 0 Success
- * \retval -1 Failure
- */
-typedef int (*ast_sip_dialog_outbound_auth_cb)(pjsip_dialog *dlg, pjsip_tx_data *tdata, void *user_data);
-
-/*!
- * \brief Set up outbound authentication on a SIP dialog
- *
- * This sets up the infrastructure so that all requests associated with a created dialog
- * can be re-sent with authentication credentials if the original request is challenged.
- *
- * \param dlg The dialog on which requests will be authenticated
- * \param endpoint The endpoint whom this dialog pertains to
- * \param cb Callback to call to send requests with authentication
- * \param user_data Data to be provided to the callback when it is called
- *
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_dialog_setup_outbound_authentication(pjsip_dialog *dlg, const struct ast_sip_endpoint *endpoint,
-		ast_sip_dialog_outbound_auth_cb cb, void *user_data);
-
-/*!
- * \brief Initialize the distributor module
- *
- * The distributor module is responsible for taking an incoming
- * SIP message and placing it into the threadpool. Once in the threadpool,
- * the distributor will perform endpoint lookups and authentication, and
- * then distribute the message up the stack to any further modules.
- *
- * \retval -1 Failure
- * \retval 0 Success
- */
-int ast_sip_initialize_distributor(void);
-
-/*!
- * \brief Destruct the distributor module.
- *
- * Unregisters pjsip modules and cleans up any allocated resources.
- */
-void ast_sip_destroy_distributor(void);
-
-/*!
- * \brief Retrieves a reference to the artificial auth.
- *
- * \retval The artificial auth
- */
-struct ast_sip_auth *ast_sip_get_artificial_auth(void);
-
-/*!
- * \brief Retrieves a reference to the artificial endpoint.
- *
- * \retval The artificial endpoint
- */
-struct ast_sip_endpoint *ast_sip_get_artificial_endpoint(void);
-
-/*!
- * \page Threading model for SIP
- *
- * There are three major types of threads that SIP will have to deal with:
- * \li Asterisk threads
- * \li PJSIP threads
- * \li SIP threadpool threads (a.k.a. "servants")
- *
- * \par Asterisk Threads
- *
- * Asterisk threads are those that originate from outside of SIP but within
- * Asterisk. The most common of these threads are PBX (channel) threads and
- * the autoservice thread. Most interaction with these threads will be through
- * channel technology callbacks. Within these threads, it is fine to handle
- * Asterisk data from outside of SIP, but any handling of SIP data should be
- * left to servants, \b especially if you wish to call into PJSIP for anything.
- * Asterisk threads are not registered with PJLIB, so attempting to call into
- * PJSIP will cause an assertion to be triggered, thus causing the program to
- * crash.
- *
- * \par PJSIP Threads
- *
- * PJSIP threads are those that originate from handling of PJSIP events, such
- * as an incoming SIP request or response, or a transaction timeout. The role
- * of these threads is to process information as quickly as possible so that
- * the next item on the SIP socket(s) can be serviced. On incoming messages,
- * Asterisk automatically will push the request to a servant thread. When your
- * module callback is called, processing will already be in a servant. However,
- * for other PSJIP events, such as transaction state changes due to timer
- * expirations, your module will be called into from a PJSIP thread. If you
- * are called into from a PJSIP thread, then you should push whatever processing
- * is needed to a servant as soon as possible. You can discern if you are currently
- * in a SIP servant thread using the \ref ast_sip_thread_is_servant function.
- *
- * \par Servants
- *
- * Servants are where the bulk of SIP work should be performed. These threads
- * exist in order to do the work that Asterisk threads and PJSIP threads hand
- * off to them. Servant threads register themselves with PJLIB, meaning that
- * they are capable of calling PJSIP and PJLIB functions if they wish.
- *
- * \par Serializer
- *
- * Tasks are handed off to servant threads using the API call \ref ast_sip_push_task.
- * The first parameter of this call is a serializer. If this pointer
- * is NULL, then the work will be handed off to whatever servant can currently handle
- * the task. If this pointer is non-NULL, then the task will not be executed until
- * previous tasks pushed with the same serializer have completed. For more information
- * on serializers and the benefits they provide, see \ref ast_threadpool_serializer
- *
- * \note
- *
- * Do not make assumptions about individual threads based on a corresponding serializer.
- * In other words, just because several tasks use the same serializer when being pushed
- * to servants, it does not mean that the same thread is necessarily going to execute those
- * tasks, even though they are all guaranteed to be executed in sequence.
- */
-
-/*!
- * \brief Create a new serializer for SIP tasks
- *
- * See \ref ast_threadpool_serializer for more information on serializers.
- * SIP creates serializers so that tasks operating on similar data will run
- * in sequence.
- *
- * \retval NULL Failure
- * \retval non-NULL Newly-created serializer
- */
-struct ast_taskprocessor *ast_sip_create_serializer(void);
-
-/*!
- * \brief Set a serializer on a SIP dialog so requests and responses are automatically serialized
- *
- * Passing a NULL serializer is a way to remove a serializer from a dialog.
- *
- * \param dlg The SIP dialog itself
- * \param serializer The serializer to use
- */
-void ast_sip_dialog_set_serializer(pjsip_dialog *dlg, struct ast_taskprocessor *serializer);
-
-/*!
- * \brief Set an endpoint on a SIP dialog so in-dialog requests do not undergo endpoint lookup.
- *
- * \param dlg The SIP dialog itself
- * \param endpoint The endpoint that this dialog is communicating with
- */
-void ast_sip_dialog_set_endpoint(pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint);
-
-/*!
- * \brief Get the endpoint associated with this dialog
- *
- * This function increases the refcount of the endpoint by one. Release
- * the reference once you are finished with the endpoint.
- *
- * \param dlg The SIP dialog from which to retrieve the endpoint
- * \retval NULL No endpoint associated with this dialog
- * \retval non-NULL The endpoint.
- */
-struct ast_sip_endpoint *ast_sip_dialog_get_endpoint(pjsip_dialog *dlg);
-
-/*!
- * \brief Pushes a task to SIP servants
- *
- * This uses the serializer provided to determine how to push the task.
- * If the serializer is NULL, then the task will be pushed to the
- * servants directly. If the serializer is non-NULL, then the task will be
- * queued behind other tasks associated with the same serializer.
- *
- * \param serializer The serializer to which the task belongs. Can be NULL
- * \param sip_task The task to execute
- * \param task_data The parameter to pass to the task when it executes
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data);
-
-/*!
- * \brief Push a task to SIP servants and wait for it to complete
- *
- * Like \ref ast_sip_push_task except that it blocks until the task completes.
- *
- * \warning \b Never use this function in a SIP servant thread. This can potentially
- * cause a deadlock. If you are in a SIP servant thread, just call your function
- * in-line.
- *
- * \param serializer The SIP serializer to which the task belongs. May be NULL.
- * \param sip_task The task to execute
- * \param task_data The parameter to pass to the task when it executes
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data);
-
-/*!
- * \brief Determine if the current thread is a SIP servant thread
- *
- * \retval 0 This is not a SIP servant thread
- * \retval 1 This is a SIP servant thread
- */
-int ast_sip_thread_is_servant(void);
-
-/*!
- * \brief SIP body description
- *
- * This contains a type and subtype that will be added as
- * the "Content-Type" for the message as well as the body
- * text.
- */
-struct ast_sip_body {
-	/*! Type of the body, such as "application" */
-	const char *type;
-	/*! Subtype of the body, such as "sdp" */
-	const char *subtype;
-	/*! The text to go in the body */
-	const char *body_text;
-};
-
-/*!
- * \brief General purpose method for creating a dialog with an endpoint
- *
- * \param endpoint A pointer to the endpoint
- * \param aor_name Optional name of the AOR to target, may even be an explicit SIP URI
- * \param request_user Optional user to place into the target URI
- *
- * \retval non-NULL success
- * \retval NULL failure
- */
- pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *aor_name, const char *request_user);
-
-/*!
- * \brief General purpose method for creating a SIP request
- *
- * Its typical use would be to create one-off requests such as an out of dialog
- * SIP MESSAGE.
- *
- * The request can either be in- or out-of-dialog. If in-dialog, the
- * dlg parameter MUST be present. If out-of-dialog the endpoint parameter
- * MUST be present. If both are present, then we will assume that the message
- * is to be sent in-dialog.
- *
- * The uri parameter can be specified if the request should be sent to an explicit
- * URI rather than one configured on the endpoint.
- *
- * \param method The method of the SIP request to send
- * \param dlg Optional. If specified, the dialog on which to request the message.
- * \param endpoint Optional. If specified, the request will be created out-of-dialog
- * to the endpoint.
- * \param uri Optional. If specified, the request will be sent to this URI rather
- * this value.
- * than one configured for the endpoint.
- * \param[out] tdata The newly-created request
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
-		struct ast_sip_endpoint *endpoint, const char *uri,
-		pjsip_tx_data **tdata);
-
-/*!
- * \brief General purpose method for sending a SIP request
- *
- * This is a companion function for \ref ast_sip_create_request. The request
- * created there can be passed to this function, though any request may be
- * passed in.
- *
- * This will automatically set up handling outbound authentication challenges if
- * they arrive.
- *
- * \param tdata The request to send
- * \param dlg Optional. If specified, the dialog on which the request should be sent
- * \param endpoint Optional. If specified, the request is sent out-of-dialog to the endpoint.
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint);
-
-/*!
- * \brief Determine if an incoming request requires authentication
- *
- * This calls into the registered authenticator's requires_authentication callback
- * in order to determine if the request requires authentication.
- *
- * If there is no registered authenticator, then authentication will be assumed
- * not to be required.
- *
- * \param endpoint The endpoint from which the request originates
- * \param rdata The incoming SIP request
- * \retval non-zero The request requires authentication
- * \retval 0 The request does not require authentication
- */
-int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
-
-/*!
- * \brief Method to determine authentication status of an incoming request
- *
- * This will call into a registered authenticator. The registered authenticator will
- * do what is necessary to determine whether the incoming request passes authentication.
- * A tentative response is passed into this function so that if, say, a digest authentication
- * challenge should be sent in the ensuing response, it can be added to the response.
- *
- * \param endpoint The endpoint from the request was sent
- * \param rdata The request to potentially authenticate
- * \param tdata Tentative response to the request
- * \return The result of checking authentication.
- */
-enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
-		pjsip_rx_data *rdata, pjsip_tx_data *tdata);
-
-/*!
- * \brief Create a response to an authentication challenge
- *
- * This will call into an outbound authenticator's create_request_with_auth callback
- * to create a new request with authentication credentials. See the create_request_with_auth
- * callback in the \ref ast_sip_outbound_authenticator structure for details about
- * the parameters and return values.
- */
-int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
-		pjsip_transaction *tsx, pjsip_tx_data **new_request);
-
-/*!
- * \brief Determine the endpoint that has sent a SIP message
- *
- * This will call into each of the registered endpoint identifiers'
- * identify_endpoint() callbacks until one returns a non-NULL endpoint.
- * This will return an ao2 object. Its reference count will need to be
- * decremented when completed using the endpoint.
- *
- * \param rdata The inbound SIP message to use when identifying the endpoint.
- * \retval NULL No matching endpoint
- * \retval non-NULL The matching endpoint
- */
-struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata);
-
-/*!
- * \brief Add a header to an outbound SIP message
- *
- * \param tdata The message to add the header to
- * \param name The header name
- * \param value The header value
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value);
-
-/*!
- * \brief Add a body to an outbound SIP message
- *
- * If this is called multiple times, the latest body will replace the current
- * body.
- *
- * \param tdata The message to add the body to
- * \param body The message body to add
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body);
-
-/*!
- * \brief Add a multipart body to an outbound SIP message
- *
- * This will treat each part of the input array as part of a multipart body and
- * add each part to the SIP message.
- *
- * \param tdata The message to add the body to
- * \param bodies The parts of the body to add
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies);
-
-/*!
- * \brief Append body data to a SIP message
- *
- * This acts mostly the same as ast_sip_add_body, except that rather than replacing
- * a body if it currently exists, it appends data to an existing body.
- *
- * \param tdata The message to append the body to
- * \param body The string to append to the end of the current body
- * \retval 0 Success
- * \retval -1 Failure
- */
-int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text);
-
-/*!
- * \brief Copy a pj_str_t into a standard character buffer.
- *
- * pj_str_t is not NULL-terminated. Any place that expects a NULL-
- * terminated string needs to have the pj_str_t copied into a separate
- * buffer.
- *
- * This method copies the pj_str_t contents into the destination buffer
- * and NULL-terminates the buffer.
- *
- * \param dest The destination buffer
- * \param src The pj_str_t to copy
- * \param size The size of the destination buffer.
- */
-void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size);
-
-/*!
- * \brief Get the looked-up endpoint on an out-of dialog request or response
- *
- * The function may ONLY be called on out-of-dialog requests or responses. For
- * in-dialog requests and responses, it is required that the user of the dialog
- * has the looked-up endpoint stored locally.
- *
- * This function should never return NULL if the message is out-of-dialog. It will
- * always return NULL if the message is in-dialog.
- *
- * This function will increase the reference count of the returned endpoint by one.
- * Release your reference using the ao2_ref function when finished.
- *
- * \param rdata Out-of-dialog request or response
- * \return The looked up endpoint
- */
-struct ast_sip_endpoint *ast_pjsip_rdata_get_endpoint(pjsip_rx_data *rdata);
-
-/*!
- * \brief Retrieve any endpoints available to sorcery.
- *
- * \retval Endpoints available to sorcery, NULL if no endpoints found.
- */
-struct ao2_container *ast_sip_get_endpoints(void);
-
-/*!
- * \brief Retrieve relevant SIP auth structures from sorcery
- *
- * \param auths Array of sorcery IDs of auth credentials to retrieve
- * \param[out] out The retrieved auths are stored here
- */
-int ast_sip_retrieve_auths(const struct ast_sip_auth_array *auths, struct ast_sip_auth **out);
-
-/*!
- * \brief Clean up retrieved auth structures from memory
- *
- * Call this function once you have completed operating on auths
- * retrieved from \ref ast_sip_retrieve_auths
- *
- * \param auths An array of auth structures to clean up
- * \param num_auths The number of auths in the array
- */
-void ast_sip_cleanup_auths(struct ast_sip_auth *auths[], size_t num_auths);
-
-/*!
- * \brief Checks if the given content type matches type/subtype.
- *
- * Compares the pjsip_media_type with the passed type and subtype and
- * returns the result of that comparison.  The media type parameters are
- * ignored.
- *
- * \param content_type The pjsip_media_type structure to compare
- * \param type The media type to compare
- * \param subtype The media subtype to compare
- * \retval 0 No match
- * \retval -1 Match
- */
-int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype);
-
-/*!
- * \brief Send a security event notification for when an invalid endpoint is requested
- *
- * \param name Name of the endpoint requested
- * \param rdata Received message
- */
-void ast_sip_report_invalid_endpoint(const char *name, pjsip_rx_data *rdata);
-
-/*!
- * \brief Send a security event notification for when an ACL check fails
- *
- * \param endpoint Pointer to the endpoint in use
- * \param rdata Received message
- * \param name Name of the ACL
- */
-void ast_sip_report_failed_acl(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, const char *name);
-
-/*!
- * \brief Send a security event notification for when a challenge response has failed
- *
- * \param endpoint Pointer to the endpoint in use
- * \param rdata Received message
- */
-void ast_sip_report_auth_failed_challenge_response(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
-
-/*!
- * \brief Send a security event notification for when authentication succeeds
- *
- * \param endpoint Pointer to the endpoint in use
- * \param rdata Received message
- */
-void ast_sip_report_auth_success(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
-
-/*!
- * \brief Send a security event notification for when an authentication challenge is sent
- *
- * \param endpoint Pointer to the endpoint in use
- * \param rdata Received message
- * \param tdata Sent message
- */
-void ast_sip_report_auth_challenge_sent(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, pjsip_tx_data *tdata);
-
-void ast_sip_initialize_global_headers(void);
-void ast_sip_destroy_global_headers(void);
-
-int ast_sip_add_global_request_header(const char *name, const char *value, int replace);
-int ast_sip_add_global_response_header(const char *name, const char *value, int replace);
-
-int ast_sip_initialize_sorcery_global(struct ast_sorcery *sorcery);
-
-#endif /* _RES_SIP_H */