initial checkin

Check in contents of upstream 1.4.2 tarball, exclude generated files.

1 files changed, 850 insertions, 0 deletions

diff --git a/src/user.h b/src/user.h
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+/*
+    Copyright (C) 2005-2009  Michel de Boer <michel@twinklephone.com>
+
+    This program is free software; you can redistribute it and/or modify
+    it under the terms of the GNU General Public License as published by
+    the Free Software Foundation; either version 2 of the License, or
+    (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU General Public License for more details.
+
+    You should have received a copy of the GNU General Public License
+    along with this program; if not, write to the Free Software
+    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+*/
+
+// NOTE:
+// When adding attributes to t_user, make sure to add them to the
+// copy constructor too!
+
+#ifndef _H_USER
+#define _H_USER
+
+#include <string>
+#include <list>
+#include <cc++/config.h>
+#include "protocol.h"
+#include "sys_settings.h"
+#include "audio/audio_codecs.h"
+#include "sockets/url.h"
+#include "threads/mutex.h"
+#include "boost/regex.hpp"
+
+// Forward declaration
+class t_request;
+
+// Default config file name
+#define USER_CONFIG_FILE	"twinkle.cfg"
+#define USER_FILE_EXT		".cfg"
+#define USER_DIR		DIR_USER
+
+#define USER_SCHEME		"sip"
+
+#define PUBLIC_SIP_PORT(u)	phone->get_public_port_sip(u)
+#define USER_HOST(u,local_ip)	phone->get_ip_sip(u,(local_ip))
+#define LOCAL_IP		user_host
+#define LOCAL_HOSTNAME		local_hostname
+
+#define SPECIAL_PHONE_SYMBOLS	"-()/."
+
+using namespace std;
+
+enum t_hold_variant {
+	HOLD_RFC2543,	// set IP = 0.0.0.0 in c-line
+	HOLD_RFC3264	// use direction attribute to put call on-hold
+};
+
+/** SIP transport mode */
+enum t_sip_transport {
+	SIP_TRANS_UDP,	/**< SIP over UDP */
+	SIP_TRANS_TCP,	/**< SIP over TCP */
+	SIP_TRANS_AUTO	/**< UDP for small messages, TCP for large messages */
+};
+
+enum t_ext_support {
+	EXT_INVALID,
+	EXT_DISABLED,
+	EXT_SUPPORTED,
+	EXT_PREFERRED,
+	EXT_REQUIRED
+};
+
+enum t_bit_rate_type {
+	BIT_RATE_INVALID,
+	BIT_RATE_CBR,	// Constant
+	BIT_RATE_VBR,	// Variable
+	BIT_RATE_ABR	// Average
+};
+
+enum t_dtmf_transport {
+	DTMF_INBAND,
+	DTMF_RFC2833,
+	DTMF_AUTO,
+	DTMF_INFO
+};
+
+enum t_g726_packing {
+	G726_PACK_RFC3551,
+	G726_PACK_AAL2
+};
+
+struct t_number_conversion {
+	boost::regex	re;
+	string		fmt;
+	
+	string str(void) const { return re.str() + " --> " + fmt; }
+};
+
+
+class t_user {
+private:
+	string			config_filename;
+	
+	// Mutex for exclusive access to the user profile
+	mutable t_recursive_mutex	mtx_user;
+
+	/** @name USER */
+	//@{
+	// SIP user
+	/** User name (public user identity). */
+	string			name;
+	
+	/** Domain of the user. */
+	string			domain;
+	
+	/** Display name. */
+	string			display;
+
+	/**
+	 * The organization will be put in an initial INVITE and in a
+	 * 200 OK on an INVITE.
+	 */
+	string			organization;
+
+	// SIP authentication
+	/** Authentication realm. An empty realm matches with all realms. */
+	string			auth_realm;
+	
+	/** Authentication name (private user identity). */
+	string			auth_name;
+	
+	/** Authentication password (aka_k for akav1-md5 authentication) */
+	string			auth_pass;
+	
+	/** Operator variant key for akav1-md5 authentication. */
+	uint8			auth_aka_op[AKA_OPLEN];
+	
+	/** Authentication management field for akav1-md5 authentication. */
+	uint8			auth_aka_amf[AKA_AMFLEN];
+	//@}
+
+
+	// SIP SERVER
+
+	// Send all non-REGISTER requests to the outbound proxy
+	bool			use_outbound_proxy;
+	t_url			outbound_proxy;
+
+	// By default only out-of-dialog requests (including the ones the
+	// establish a dialog) are sent to the outbound proxy.
+	// In-dialog requests go to the address established via the contact
+	// header.
+	// By setting this parameter to true, in-dialog requests go to the
+	// outbound proxy as well.
+	bool			all_requests_to_proxy;
+
+	// Only send the request to the proxy, if the destination cannot
+	// be resolved to an IP address, adhearing to the previous setting
+	// though. I.e. use_outbound_proxy must be true. And an in-dialog
+	// request will only be sent to the proxy if all_requests_to_proxy
+	// is true.
+	bool			non_resolvable_to_proxy;
+
+	// Send REGISTER to registrar
+	bool			use_registrar;
+	t_url			registrar;
+	
+	// Registration time requested by the client. If set to zero, then
+	// no specific time is requested. The registrar will set a time.
+	unsigned long		registration_time;
+
+	// Automatically register at startup of the client.
+	bool			register_at_startup;
+
+	// q-value for registration
+	bool			reg_add_qvalue;
+	float			reg_qvalue;
+
+
+	// AUDIO
+
+	list<t_audio_codec>	codecs; // in order of preference
+	unsigned short		ptime; // ptime (ms) for G.711/G.726
+	
+	// For outgoing calls, obey the preference from the far-end (SDP answer),
+	// i.e. pick the first codec from the SDP answer that we support.
+	bool			out_obey_far_end_codec_pref;
+	
+	// For incoming calls, obey the preference from the far-end (SDP offer),
+	// i.e. pick the first codec from the SDP offer that we support.
+	bool			in_obey_far_end_codec_pref;
+	
+	// RTP dynamic payload types for speex
+	unsigned short		speex_nb_payload_type;
+	unsigned short		speex_wb_payload_type;
+	unsigned short		speex_uwb_payload_type;
+	
+	// Speex preprocessing options
+	bool			speex_dsp_vad;       // voice activity reduction
+	bool			speex_dsp_agc;       // automatic gain control
+	bool			speex_dsp_aec;       // acoustic echo cancellation
+	bool			speex_dsp_nrd;       // noise reduction
+	unsigned short		speex_dsp_agc_level; // gain level of AGC (1-100[%])
+
+	// Speex coding options
+	t_bit_rate_type		speex_bit_rate_type;
+	int			speex_abr_nb;
+	int			speex_abr_wb;
+	bool			speex_dtx;
+	bool			speex_penh;
+	unsigned short		speex_complexity;
+	unsigned short		speex_quality; // quality measure (worst 0-10 best)
+	
+	// RTP dynamic payload types for iLBC
+	unsigned short		ilbc_payload_type;
+	
+	// iLBC options
+	unsigned short		ilbc_mode; // 20 or 30 ms frame size
+	
+	// RTP dynamic payload types for G.726
+	unsigned short		g726_16_payload_type;
+	unsigned short		g726_24_payload_type;
+	unsigned short		g726_32_payload_type;
+	unsigned short		g726_40_payload_type;
+	
+	// Bit packing order for G,726
+	t_g726_packing		g726_packing;
+	
+	// Transport mode for DTMF
+	t_dtmf_transport	dtmf_transport;
+
+	// RTP dynamic payload type for out-of-band DTMF.
+	unsigned short		dtmf_payload_type;
+
+	// DTMF duration and pause between 2 tones. During the pause the last
+	// DTMF event will be repeated so the far end can detect the end of
+	// the event in case of packet loss.
+	unsigned short		dtmf_duration; // ms
+	unsigned short		dtmf_pause; // ms
+	
+	// Volume of the tone in -dBm
+	unsigned short		dtmf_volume;
+
+
+	/** @name SIP PROTOCOL */
+	//@{
+	// SIP protocol options
+	// hold variants: rfc2543, rfc3264
+	// rfc2543 - set IP address to 0.0.0.0
+	// rfc3264 - use direction attribute (sendrecv, sendonly, ...)
+	t_hold_variant		hold_variant;
+
+	// Indicate if the mandatory Max-Forwards header should be present.
+	// If true and the header is missing, then the request will fail.
+	bool			check_max_forwards;
+	
+	// RFC 3261 10.3 states that a registrar must include a contact
+	// header in a 200 OK on a REGISTER. This contact should match the
+	// contact that a UA puts in the REGISTER. Unfortunately many
+	// registrars do not include the contact header or put a wrong
+	// IP address in the host-part due to NAT.
+	// This settings allows for a missing/non-matching contact header.
+	// In that case Twinkle assumes that it is registered for the
+	// requested interval.
+	bool			allow_missing_contact_reg;
+
+	// Indicate the place of the requested registration time in a REGISTER.
+	// true - expires parameter in contact header
+	// false - Expires header
+	bool			registration_time_in_contact;
+
+	// Indicate if compact header names should be used in outgoing messages.
+	bool			compact_headers;
+	
+	// Indicate if headers containing multiple values should be encoded
+	// as a comma separated list or as multiple headers.
+	bool			encode_multi_values_as_list;
+	
+	// Indicate if a unique contact name should be created by using
+	// the domain name: username_domain
+	// If false then the SIP user name is used as contact name
+	bool			use_domain_in_contact;
+	
+	// Allow SDP to change in different INVITE responses. 
+	// According to RFC 3261 13.2.1, if SDP is received in a 1XX response,
+	// then SDP received in subsequent responses should be ignored.
+	// Some SIP proxies do send different SDP in 1XX and 200 though.
+	// E.g. first SDP is to play ring tone, second SDP is to create
+	// an end-to-end media path.
+	bool			allow_sdp_change;
+
+	// Redirections
+	// Allow redirection of a request when a 3XX is received.
+	bool			allow_redirection;
+
+	// Ask user for permission to redirect a request when a 3XX is received.
+	bool			ask_user_to_redirect;
+
+	// Maximum number of locations to be tried when a request is redirected.
+	unsigned short		max_redirections;
+
+	// SIP extensions
+	// 100rel extension (PRACK, RFC 3262)
+	// Possible values:
+	// - disabled	100rel extension is disabled
+	// - supported	100rel is supported (it is added in the supported header of
+	//               an outgoing INVITE). A far-end can now require a PRACK on a
+	//               1xx response.
+	// - required	100rel is required (it is put in the require header of an
+	//               outgoing INVITE).
+	//		If an incoming INVITE indicates that it supports 100rel, then
+	//		Twinkle will require a PRACK when sending a 1xx response.
+	// - preferred	Similar to required, but if a call fails because the far-end
+	//		indicates it does not support 100rel (420 response) then the
+	//		call will be re-attempted without the 100rel requirement.
+	t_ext_support		ext_100rel;
+	
+	// Replaces (RFC 3891)
+	bool			ext_replaces;
+	//@}
+
+	/** @name REFER options */
+	//@{
+	/** Hold the current call when an incoming REFER is accepted. */
+	bool			referee_hold;
+
+	/** Hold the current call before sending a REFER. */
+	bool			referrer_hold;
+
+	/** Allow an incoming refer */
+	bool			allow_refer;
+
+	/** Ask user for permission when a REFER is received. */
+	bool			ask_user_to_refer;
+
+	/** Referrer automatically refreshes subscription before expiry. */
+	bool			auto_refresh_refer_sub;
+	
+	/**
+	 * An attended transfer should use the contact-URI of the transfer target.
+	 * This contact-URI is not always globally routable however. As an
+	 * alternative the AoR (address of record) can be used. Disadvantage is
+	 * that the AoR may route to multiple phones in case of forking, whereas
+	 * the contact-URI routes to a particular phone.
+	 */
+	bool			attended_refer_to_aor;
+	
+	/** 
+	 * Allow to transfer a call while the consultation call is still
+	 * in progress.
+	 */
+	bool			allow_transfer_consultation_inprog;
+	//@}
+	
+	/** @name Privacy options */
+	//@{
+	// Send P-Preferred-Identity header in initial INVITE when hiding
+	// user identity.
+	bool			send_p_preferred_id;
+	//@}
+	
+	/** @name Transport */
+	//@{
+	/** SIP transport protocol */
+	t_sip_transport		sip_transport;
+	
+	/** 
+	 * Threshold to decide which transport to use in auto transport mode. 
+	 * A message with a size up to this threshold is sent via UDP. Larger messages
+	 * are sent via TCP.
+	 */
+	unsigned short		sip_transport_udp_threshold;
+	//@}
+
+	/** @name NAT */
+	//@{
+	/**
+	 * NAT traversal
+	 * You can set nat_public_ip to your public IP or FQDN if you are behind
+	 * a NAT. This will then be used inside the SIP messages instead of your
+	 * private IP. On your NAT you have to create static bindings for port 5060
+	 * and ports 8000 - 8005 to the same ports on your private IP address.
+	 */
+	bool			use_nat_public_ip;
+	
+	/** The public IP address of the NAT device. */
+	string			nat_public_ip;
+	
+	/** NAT traversal via STUN. */
+	bool			use_stun;
+	
+	/** URL of the STUN server. */
+	t_url			stun_server;
+	
+	/** User persistent TCP connections. */
+	bool			persistent_tcp;
+	
+	/** Enable sending of NAT keepalive packets for UDP. */
+	bool			enable_nat_keepalive;
+	//@}
+
+	/** @name TIMERS */
+	//@{
+	/** 
+	 * Noanswer timer is started when an initial INVITE is received. If
+	 * the user does not respond within the timer, then the call will be
+	 * released with a 480 Temporarily Unavailable response.
+	 */
+	unsigned short		timer_noanswer; // seconds
+	
+	/** Duration of NAT keepalive timer (s) */
+	unsigned short		timer_nat_keepalive;
+	
+	/** Duration of TCP ping timer (s) */
+	unsigned short		timer_tcp_ping;
+	//@}
+
+	/** @name ADDRESS FORMAT */
+	//@{
+	/**
+	 * Telephone numbers
+	 * Display only the user-part of a URI if it is a telephone number
+	 * I.e. the user=phone parameter is present, or the user indicated
+	 * that the format of the user-part is a telephone number.
+	 * If the URI is a tel-URI then display the telephone number.
+	 */
+	bool			display_useronly_phone;
+
+	/**
+	 * Consider user-parts that consist of 0-9,+,-,*,# as a telephone
+	 * number. I.e. in outgoing messages the user=phone parameter will
+	 * be added to the URI. For incoming messages the URI will be considered
+	 * to be a telephone number regardless of the presence of the
+	 * user=phone parameter.
+	 */
+	bool			numerical_user_is_phone;
+	
+	/** Remove special symbols from numerical dial strings */
+	bool			remove_special_phone_symbols;
+	
+	/** Special symbols that must be removed from telephone numbers */
+	string			special_phone_symbols;
+	
+	/**
+	 * If the user enters a telephone number as address, then complete it
+	 * to a tel-URI instead of a sip-URI.
+	 */
+	bool			use_tel_uri_for_phone;
+	
+	/** Number conversion */
+	list<t_number_conversion>	number_conversions;
+	//@}
+	
+	// RING TONES
+	string		ringtone_file;
+	string		ringback_file;
+	
+	// SCRIPTS
+	// Script to be called on incoming call
+	string		script_incoming_call;
+	string		script_in_call_answered;
+	string		script_in_call_failed;
+	string		script_outgoing_call;
+	string		script_out_call_answered;
+	string		script_out_call_failed;
+	string		script_local_release;
+	string		script_remote_release;
+	
+	// SECURITY
+	// zrtp setting
+	bool		zrtp_enabled;
+	
+	// Popup warning when far-end sends goclear command
+	bool		zrtp_goclear_warning;
+	
+	// Send a=zrtp in SDP
+	bool		zrtp_sdp;
+	
+	// Only negotiate zrtp if far-end signalled support for zrtp
+	bool		zrtp_send_if_supported;
+	
+	// MWI
+	// Indicate if MWI is sollicited or unsollicited.
+	// RFC 3842 specifies that MWI must be sollicited (SUBSCRIBE).
+	// Asterisk however only supported non-standard unsollicited MWI.
+	bool		mwi_sollicited;
+	
+	// User name for subscribing to the mailbox
+	string		mwi_user;
+	
+	// The mailbox server to which the SUBSCRIBE must be sent
+	t_url		mwi_server;
+	
+	// Send the SUBSCRIBE via the proxy to the mailbox server
+	bool		mwi_via_proxy;
+	
+	// Requested MWI subscription duration
+	unsigned long	mwi_subscription_time;
+	
+	// The voice mail address to call to access messages
+	string		mwi_vm_address;
+	
+	/** @name INSTANT MESSAGE */
+	//@{
+	/** Maximum number of simultaneous IM sessions. */
+	unsigned short	im_max_sessions;
+	
+	/** Flag to indicate that IM is-composing indications (RFC 3994) should be sent. */
+	bool		im_send_iscomposing;
+	//@}
+	
+	/** @name PRESENCE */
+	//@{
+	/** Requested presence subscription duration in seconds. */
+	unsigned long	pres_subscription_time;
+	
+	/** Requested presence publication duration in seconds. */
+	unsigned long	pres_publication_time;
+	
+	/** Publish online presence state at startup */
+	bool		pres_publish_startup;
+	//@}
+
+	t_ext_support str2ext_support(const string &s) const;
+	string ext_support2str(t_ext_support e) const;
+	t_bit_rate_type str2bit_rate_type(const string &s) const;
+	string bit_rate_type2str(t_bit_rate_type b) const;
+	t_dtmf_transport str2dtmf_transport(const string &s) const;
+	string dtmf_transport2str(t_dtmf_transport d) const;
+	t_g726_packing str2g726_packing(const string &s) const;
+	string g726_packing2str(t_g726_packing packing) const;
+	t_sip_transport str2sip_transport(const string &s) const;
+	string sip_transport2str(t_sip_transport transport) const;
+	
+	// Parse a number conversion rule
+	// If the rule can be parsed, then c contains the conversion rule and
+	// true is returned. Otherwise false is returned.
+	bool parse_num_conversion(const string &value, t_number_conversion &c);
+	
+	// Set a server URL.
+	// Returns false, if the passed value is not a valid URL.
+	bool set_server_value(t_url &server, const string &scheme, const string &value);
+	
+public:
+	t_user();
+	t_user(const t_user &u);
+	
+	t_user *copy(void) const;
+	
+	/** @name Getters */
+	//@{
+	string get_name(void) const;
+	string get_domain(void) const;
+	string get_display(bool anonymous) const;	
+	string get_organization(void) const;
+	string get_auth_realm(void) const;
+	string get_auth_name(void) const;
+	string get_auth_pass(void) const;
+	void get_auth_aka_op(uint8 *aka_op) const;
+	void get_auth_aka_amf(uint8 *aka_amf) const;
+	bool get_use_outbound_proxy(void) const;
+	t_url get_outbound_proxy(void) const;
+	bool get_all_requests_to_proxy(void) const;
+	bool get_non_resolvable_to_proxy(void) const;
+	bool get_use_registrar(void) const;
+	t_url get_registrar(void) const;
+	unsigned long get_registration_time(void) const;
+	bool get_register_at_startup(void) const;
+	bool get_reg_add_qvalue(void) const;
+	float get_reg_qvalue(void) const;
+	list<t_audio_codec> get_codecs(void) const;
+	unsigned short get_ptime(void) const;
+	bool get_out_obey_far_end_codec_pref(void) const;
+	bool get_in_obey_far_end_codec_pref(void) const;
+	unsigned short get_speex_nb_payload_type(void) const;
+	unsigned short get_speex_wb_payload_type(void) const;
+	unsigned short get_speex_uwb_payload_type(void) const;
+	t_bit_rate_type get_speex_bit_rate_type(void) const;
+	int get_speex_abr_nb(void) const;
+	int get_speex_abr_wb(void) const;
+	bool get_speex_dtx(void) const;
+	bool get_speex_penh(void) const;
+	unsigned short get_speex_quality(void) const;
+	unsigned short get_speex_complexity(void) const;
+	bool get_speex_dsp_vad(void) const;
+	bool get_speex_dsp_agc(void) const;
+	bool get_speex_dsp_aec(void) const;
+	bool get_speex_dsp_nrd(void) const;
+	unsigned short get_speex_dsp_agc_level(void) const;
+	unsigned short get_ilbc_payload_type(void) const;
+	unsigned short get_ilbc_mode(void) const;
+	unsigned short get_g726_16_payload_type(void) const;
+	unsigned short get_g726_24_payload_type(void) const;
+	unsigned short get_g726_32_payload_type(void) const;
+	unsigned short get_g726_40_payload_type(void) const;
+	t_g726_packing get_g726_packing(void) const;
+	t_dtmf_transport get_dtmf_transport(void) const;
+	unsigned short get_dtmf_payload_type(void) const;
+	unsigned short get_dtmf_duration(void) const;
+	unsigned short get_dtmf_pause(void) const;
+	unsigned short get_dtmf_volume(void) const;
+	t_hold_variant get_hold_variant(void) const;
+	bool get_check_max_forwards(void) const;
+	bool get_allow_missing_contact_reg(void) const;
+	bool get_registration_time_in_contact(void) const;
+	bool get_compact_headers(void) const;
+	bool get_encode_multi_values_as_list(void) const;
+	bool get_use_domain_in_contact(void) const;
+	bool get_allow_sdp_change(void) const;
+	bool get_allow_redirection(void) const;
+	bool get_ask_user_to_redirect(void) const;
+	unsigned short get_max_redirections(void) const;
+	t_ext_support get_ext_100rel(void) const;
+	bool get_ext_replaces(void) const;
+	bool get_referee_hold(void) const;
+	bool get_referrer_hold(void) const;
+	bool get_allow_refer(void) const;
+	bool get_ask_user_to_refer(void) const;
+	bool get_auto_refresh_refer_sub(void) const;
+	bool get_attended_refer_to_aor(void) const;
+	bool get_allow_transfer_consultation_inprog(void) const;
+	bool get_send_p_preferred_id(void) const;
+	t_sip_transport get_sip_transport(void) const;
+	unsigned short get_sip_transport_udp_threshold(void) const;
+	bool get_use_nat_public_ip(void) const;
+	string get_nat_public_ip(void) const;
+	bool get_use_stun(void) const;
+	t_url get_stun_server(void) const;
+	bool get_persistent_tcp(void) const;
+	bool get_enable_nat_keepalive(void) const;
+	unsigned short get_timer_noanswer(void) const;
+	unsigned short get_timer_nat_keepalive(void) const; 
+	unsigned short get_timer_tcp_ping(void) const;
+	bool get_display_useronly_phone(void) const;
+	bool get_numerical_user_is_phone(void) const;
+	bool get_remove_special_phone_symbols(void) const;
+	string get_special_phone_symbols(void) const;
+	bool get_use_tel_uri_for_phone(void) const;
+	string get_ringtone_file(void) const;
+	string get_ringback_file(void) const;
+	string get_script_incoming_call(void) const;
+	string get_script_in_call_answered(void) const;
+	string get_script_in_call_failed(void) const;
+	string get_script_outgoing_call(void) const;
+	string get_script_out_call_answered(void) const;
+	string get_script_out_call_failed(void) const;
+	string get_script_local_release(void) const;
+	string get_script_remote_release(void) const;
+	list<t_number_conversion> get_number_conversions(void) const;
+	bool get_zrtp_enabled(void) const;
+	bool get_zrtp_goclear_warning(void) const;
+	bool get_zrtp_sdp(void) const;
+	bool get_zrtp_send_if_supported(void) const;
+	bool get_mwi_sollicited(void) const;
+	string get_mwi_user(void) const;
+	t_url get_mwi_server(void) const;
+	bool get_mwi_via_proxy(void) const;
+	unsigned long get_mwi_subscription_time(void) const;
+	string get_mwi_vm_address(void) const;
+	unsigned short get_im_max_sessions(void) const;
+	bool get_im_send_iscomposing(void) const;
+	unsigned long get_pres_subscription_time(void) const;
+	unsigned long get_pres_publication_time(void) const;
+	bool get_pres_publish_startup(void) const;
+	//@}
+
+	
+	/** @name Setters */
+	//@{
+	void set_name(const string &_name);
+	void set_domain(const string &_domain);
+	void set_display(const string &_display);	
+	void set_organization(const string &_organization);
+	void set_auth_realm(const string &realm);
+	void set_auth_name(const string &name);
+	void set_auth_pass(const string &pass);
+	void set_auth_aka_op(const uint8 *aka_op);
+	void set_auth_aka_amf(const uint8 *aka_amf);
+	void set_use_outbound_proxy(bool b);
+	void set_outbound_proxy(const t_url &url);
+	void set_all_requests_to_proxy(bool b);
+	void set_non_resolvable_to_proxy(bool b);
+	void set_use_registrar(bool b);
+	void set_registrar(const t_url &url);
+	void set_registration_time(const unsigned long time);
+	void set_register_at_startup(bool b);
+	void set_reg_add_qvalue(bool b);
+	void set_reg_qvalue(float q);
+	void set_codecs(const list<t_audio_codec> &_codecs);
+	void set_ptime(unsigned short _ptime);
+	void set_out_obey_far_end_codec_pref(bool b);
+	void set_in_obey_far_end_codec_pref(bool b);
+	void set_speex_nb_payload_type(unsigned short payload_type);
+	void set_speex_wb_payload_type(unsigned short payload_type);
+	void set_speex_uwb_payload_type(unsigned short payload_type);
+	void set_speex_bit_rate_type(t_bit_rate_type bit_rate_type);
+	void set_speex_abr_nb(int abr);
+	void set_speex_abr_wb(int abr);
+	void set_speex_dtx(bool b);
+	void set_speex_penh(bool b);
+	void set_speex_quality(unsigned short quality);
+	void set_speex_complexity(unsigned short complexity);
+	void set_speex_dsp_vad(bool b);
+	void set_speex_dsp_agc(bool b);
+	void set_speex_dsp_aec(bool b);
+	void set_speex_dsp_nrd(bool b);
+	void set_speex_dsp_agc_level(unsigned short level);
+	void set_ilbc_payload_type(unsigned short payload_type);
+	void set_g726_16_payload_type(unsigned short payload_type);
+	void set_g726_24_payload_type(unsigned short payload_type);
+	void set_g726_32_payload_type(unsigned short payload_type);
+	void set_g726_40_payload_type(unsigned short payload_type);
+	void set_g726_packing(t_g726_packing packing);
+	void set_ilbc_mode(unsigned short mode);
+	void set_dtmf_transport(t_dtmf_transport _dtmf_transport);
+	void set_dtmf_payload_type(unsigned short payload_type);
+	void set_dtmf_duration(unsigned short duration);
+	void set_dtmf_pause(unsigned short pause);
+	void set_dtmf_volume(unsigned short volume);
+	void set_hold_variant(t_hold_variant _hold_variant);
+	void set_check_max_forwards(bool b);
+	void set_allow_missing_contact_reg(bool b);
+	void set_registration_time_in_contact(bool b);
+	void set_compact_headers(bool b);
+	void set_encode_multi_values_as_list(bool b);
+	void set_use_domain_in_contact(bool b);
+	void set_allow_sdp_change(bool b);
+	void set_allow_redirection(bool b);
+	void set_ask_user_to_redirect(bool b);
+	void set_max_redirections(unsigned short _max_redirections);
+	void set_ext_100rel(t_ext_support ext_support);
+	void set_ext_replaces(bool b);
+	void set_referee_hold(bool b);
+	void set_referrer_hold(bool b);
+	void set_allow_refer(bool b);
+	void set_ask_user_to_refer(bool b);
+	void set_auto_refresh_refer_sub(bool b);
+	void set_attended_refer_to_aor(bool b);
+	void set_allow_transfer_consultation_inprog(bool b);
+	void set_send_p_preferred_id(bool b);
+	void set_sip_transport(t_sip_transport transport);
+	void set_sip_transport_udp_threshold(unsigned short threshold);
+	void set_use_nat_public_ip(bool b);
+	void set_nat_public_ip(const string &public_ip);
+	void set_use_stun(bool b);
+	void set_stun_server(const t_url &url);
+	void set_persistent_tcp(bool b);
+	void set_enable_nat_keepalive(bool b);
+	void set_timer_noanswer(unsigned short timer);
+	void set_timer_nat_keepalive(unsigned short timer); 
+	void set_timer_tcp_ping(unsigned short timer);
+	void set_display_useronly_phone(bool b);
+	void set_numerical_user_is_phone(bool b);
+	void set_remove_special_phone_symbols(bool b);
+	void set_special_phone_symbols(const string &symbols);
+	void set_use_tel_uri_for_phone(bool b);
+	void set_ringtone_file(const string &file);
+	void set_ringback_file(const string &file);
+	void set_script_incoming_call(const string &script);
+	void set_script_in_call_answered(const string &script);
+	void set_script_in_call_failed(const string &script);
+	void set_script_outgoing_call(const string &script);
+	void set_script_out_call_answered(const string &script);
+	void set_script_out_call_failed(const string &script);
+	void set_script_local_release(const string &script);
+	void set_script_remote_release(const string &script);
+	void set_number_conversions(const list<t_number_conversion> &l);
+	void set_zrtp_enabled(bool b);
+	void set_zrtp_goclear_warning(bool b);
+	void set_zrtp_sdp(bool b);
+	void set_zrtp_send_if_supported(bool b);
+	void set_mwi_sollicited(bool b);
+	void set_mwi_user(const string &user);
+	void set_mwi_server(const t_url &url);
+	void set_mwi_via_proxy(bool b);
+	void set_mwi_subscription_time(unsigned long t);
+	void set_mwi_vm_address(const string &address);
+	void set_im_max_sessions(unsigned short max_sessions);
+	void set_im_send_iscomposing(bool b);
+	void set_pres_subscription_time(unsigned long t);
+	void set_pres_publication_time(unsigned long t);
+	void set_pres_publish_startup(bool b);
+	//@}
+
+	// Read and parse a config file into the user object.
+	// Returns false if it fails. error_msg is an error message that can
+	// be given to the user.
+	bool read_config(const string &filename, string &error_msg);
+
+	/**
+	 * Write the settings into a config file.
+	 * @param filename [in] Name of the file to write.
+	 * @param error_msg [out] Human readable error message when writing fails.
+	 * @return Returns true of writing succeeded, otherwise false.
+	 */
+	bool write_config(const string &filename, string &error_msg);
+	
+	/** Get the file name for this user profile */
+	string get_filename(void) const;
+
+	/** 
+	 * Set a config file name.
+	 * @return True if file name did not yet exist.
+	 * @return False if file name already exists.
+	 */
+	bool set_config(string _filename);
+
+	// Get the name of the profile (filename without extension)
+	string get_profile_name(void) const;
+	
+	// Expand file name to a fully qualified file name
+	string expand_filename(const string &filename);
+	
+	// The contact name is created from the name and domain values.
+	// Just the name value is not unique when multiple user profiles are
+	// activated.
+	string get_contact_name(void) const;
+	
+	// Returns "display <sip:name@domain>"
+	string get_display_uri(void) const;
+	
+	// Check if all required extensions are supported
+	bool check_required_ext(t_request *r, list<string> &unsupported) const;
+	
+	/**
+	 * Create contact URI.
+	 * @param anonymous [in] Indicates if an anonymous contact should be created.
+	 * @param auto_ip [in] Automatically determined local IP address that should be
+	 *                     used if not IP address has been determined through other means.
+	 * @return String representation of the contact URI.
+	 */
+	string create_user_contact(bool anonymous, const string &auto_ip);
+	
+	// Create user uri
+	string create_user_uri(bool anonymous);
+	
+	// Convert a number by applying the number conversions.
+	string convert_number(const string &number, const list<t_number_conversion> &l) const;
+	string convert_number(const string &number) const;
+	
+	// Get URI for sending a SUBSCRIBE for MWI
+	t_url get_mwi_uri(void) const;
+	
+	/** Is this a user profile for a Diamondcard account? */
+	bool is_diamondcard_account(void) const;
+};
+
+#endif