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Last updated: Sat Feb 3 05:00:41 2007

Asterisk developer's documentation :: Codename Pineapple

---

chan_sip3.c

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/* CODENAME PINEAPPLE - THIS IS VERY EXPERIMENTAL. 
   IF YOU USE THIS IN PRODUCTION, I WILL NOT SUPPORT YOU...
*
*  -- Mail bugs to oej@edvina.net, do not file them in the bug tracker
*/

/* To skip to source code, search for "-END-" */

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2007, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 * Chan_sip3 changes by Olle E. Johansson <oej@edvina.net>
 *
 *
 *
 * 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
 * Version 3 of chan_sip
 *
 * \author Mark Spencer <markster@digium.com>
 * \author Olle E. Johansson <oej@edvina.net> (all the chan_sip3 changes)
 *
 * See Also:
 * \arg \ref AstCREDITS
 *
 * Implementation of RFC 3261 - without S/MIME, TCP and TLS support
 * Configuration file \link Config_sip sip.conf \endlink
 *
 *
 * \todo SIP over TCP
 * \todo SIP over TLS
 * \todo Better support of forking
 * \todo VIA branch tag transaction checking
 * \todo Transaction support
 *
 * \ingroup channel_drivers
 */

/*! \page chan_sip3_overview Chan_SIP3:: Overview
 *
 * \par Overview of the handling of SIP sessions
 * The SIP channel handles several types of SIP sessions, or dialogs,
 * not all of them being "telephone calls".
 * - Incoming calls that will be sent to the PBX core
 * - Outgoing calls, generated by the PBX
 * - SIP subscriptions and notifications of states and voicemail messages
 * - SIP registrations, both inbound and outbound
 * - SIP peer management (peerpoke, OPTIONS)
 * - SIP text messages
 *
 * In the SIP channel, there's a list of active SIP dialogs, which includes
 * all of these when they are active. "sip show channels" in the CLI will
 * show most of these, excluding subscriptions which are shown by
 * "sip show subscriptions"
 *
 * \par incoming packets
 * Incoming packets are received in the monitoring thread, then handled by
 * sipsock_read(). This function parses the packet and matches an existing
 * dialog or starts a new SIP dialog.
 * 
 * sipsock_read sends the packet to handle_request(), that parses a bit more.
 * if it's a response to an outbound request, it's sent to handle_response().
 * If it is a request, handle_request sends it to one of a list of functions
 * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
 * sipsock_read locks the ast_channel if it exists (an active call) and
 * unlocks it after we have processed the SIP message.
 *
 * A new INVITE is sent to handle_request_invite(), that will end up
 * starting a new channel in the PBX, the new channel after that executing
 * in a separate channel thread. This is an incoming "call".
 * When the call is answered, either by a bridged channel or the PBX itself
 * the sip_answer() function is called.
 *
 * The actual media - Video or Audio - is mostly handled by the RTP subsystem
 * in rtp.c 
 * 
 * \par Outbound calls
 * Outbound calls are set up by the PBX through the sip_request_call()
 * function. After that, they are activated by sip_call().
 * 
 * \par Hanging up
 * The PBX issues a hangup on both incoming and outgoing calls through
 * the sip_hangup() function
 *
 * \section SIP3dialog Dialogs, packets, transactions
 *
 * This version of chan_sip has a concept of a SIP dialog (\ref sip_dialog) that
 * lives throughut a session - a call (INVITE), a registration (REGISTER)
 * or a subscription for status (SUBSCRIBE)... We do save new requests we
 * send in a queue (sip_dialog->packets) until they're acknowledged by the
 * remote party, when we remove them (sip_ack() )
 * 
 * For requests and responses, we store them in a structure called \ref sip_request
 * and process them. The matching between a response or a new request and an
 * existing dialog is done in match_or_create_dialog().
 *
 */

/*!   \page chan_sip3_00index Chan_sip3: Index over docs
   \title Chan_sip3 :: Index

   - \ref chan_sip3_overview
   - \subpage chan_sip3_start
   - \subpage chan_sip3_security
   - \subpage chan_sip3_objects
   - \subpage chan_sip3_registrydb
   - \subpage chan_sip3_files
   - \subpage chan_sip3_auth
   - \subpage chan_sip3_dialogs
   - \subpage chan_sip3_transactions
   - \subpage sip3_timer_doc
   - \subpage sip3_dialog_match
   - \subpage chan_sip3_dialstring
   - \subpage chan_sip3_natsupport

   \par todo Things to do, ideas
   - \subpage chan_sip3_todo
   - \subpage chan_sip3_subs

*/

/*! 
   \page chan_sip3_start Chan_sip3: Welcome to Codename Pineapple !

   \title Chan_sip3: What's this?
   This is a re-work of the SIP channel in Asterisk. 
   This channel will not be backwards compatible with the old 
   sip channel. In order to be more SIP compatible, I will have
   to break the backwards compatibility. That's why the old channel
   will still be around for a while.

   Chan_sip3 is the road towards security (see \ref chan_sip3_security).
   By adding a transaction layer and support for TCP connections, we can
   add TLS for the TCP connections and negotiate keys for secure media
   with SRTP.

   ** This work is sponsored by voop.com - the Internet Dialtone.
      I am open for more sponsors - contact me on oej@edvina.net

   \page chan_sip3_todo Chan_sip3: Things to do
   \b Done
   - Trying to reduce memory allocations for packets.
      - sipsock_read allocates a packet that stays in memory
        until the transaction is finished. If it's an initial
        request, it's flagged to stay in memory and kept until
        destruction of the dialog (or replacement of initreq).
      - The issue here is parsing, since parsing destroys the
        in-memory copy of the outbound message thus stopping
        proper re-transmits. Added flag for parsing of packet,
        trying to delay parsing until we send a response.
   - Added new CLI command "sip list configs" to list all configuration options 
     Mostly for debugging

   - Added new configuration engine
   - Add T1 timer configuration settings
   - Added configurable T2 timer, see \ref sip3_timer_doc
   - Added time to astdb registry storage, so that expired registrations
     won't be activated at restart
   - removed pedantic mode
   - added config option for qualify frequency timers
   - merged peermatch and sipregister branches
   - removed "type=user"
   - change "sip nodebug" to "sip debug off" and "sip debug" to "sip debug on" - done
   - change "sip history" and "nohistory" to "on/off" - done
   - "sip show/list peers" is now "sip show/list phones"
   - manager command renamed - SIPdevices and SIPshowdevice
   - Added "authuser" configuration option for trunks and services
   - Added "domain" configuration option for all devices
   - Fixed handling of too short registration times (sending 423)
   - T38 does no longer depend on canreinvite settings
   - removed userconf support (in favour of astum)

   \b Larger changes required outside of chan_sip
   - dnsmgr needs to follow DNS ttl
   - dnsmgr needs to handle SRV, NAPTR

   \b Halfdone
   - Added separate TOS setting for presence. Need to run setsockopt
     in a locked socket for that to work on the SIP interface.

   \par Todo - architecture
   - check if the "defaultport" and "port" settings are working - port for remote peer?
   - dnsmgr needs to be integrated and updated
   - netsock?
   - thread and separate port for outbound registrations
   - receive queue between sipsock_read and handle_request
   - inbound call authentication

   \par Todo - ideas
   - Implement support for a:rtcp sdp (needs changes in the rtp interface)
   - Implement "busylimit" for signalling busy, but not enforcing a call limit
   - Use "accept-language" to set language tags in error messages etc
   - Implement "holdaction = music | sendhold"
   - Handle 423 Interval too brief on registrations
   - Accept-language to language tag.
   - Fix T4 implementation
      - Configuration setting implemented in global
   - Check if usereqphone is a global flag
   - Fix compact headers per peer
   - Always enable "alwaysauthreject" and remove that option
   - Check "insecure" option - do we still need it?
   - Only check for pickup code if callgroup/pickupgroups are specified in config
   - check resp 491 to INVITE processing
   - Make show devices and the completion support domains too
   - Fix realtime caching and optional loading
   - Clean up the authuser/username/peername mess!
      - authuser as a separate config option, please, please
   - Split up source code file
   - Add astum
   - Add auto-nat for RFC 1918 networks
   - Add type=device for peers
   - Add type=service for register= replacement
   - Add type=trunk definition, based on domain routing
   - Implement state engine for dialogs
   - Implement transactions
   - Implement state engine for transactions
   - Implement real realtime caching
   - Implement realtime static loading for MWI and qualify support
   - Implement remote MWI notification
   - Implement remote subscriptions
   - Implement improved SIP domain support
      - Document these ideas!
   - Prove transaction engine by implementing PRACK
   - Implement netsock API in this channel
   - Add File's multithreading code
   - Make debugaddr a ha list instead of one address and move it out of sipnet
   - Save the last sent request/response for re-transmits
   - RTP keepalives (STUN) for video
   - Clean up 302 redirect - remove "promisredir" setting 

   \b Maybe
   - add support for Path header
      the Path is arriving with Register requests, saved in location
      and used as a Route: header in the outbound request
   - Add support for the "norefersub" option
   - Add support for GRUU

   - ... And much more
*/
/*!
   \page chan_sip3_security   Chan_sip3: The road to SIP security

   Codename pineapple is the road towards SIP security. 

   - SIP/TLS is the way to secure signalling
   - In order to get there, we need TCP
   - In order to get TCP, we need a transaction state engine
   - We also need a separation of network transmission over
      reliable and unreliable transports and the SIP
      core
   - When we have that, we can add SRTP

 */
/*!
   \page chan_sip3_objects Chan_sip3: Devices, trunks and services
   - \b phones are devices that connect to Asterisk. They register with  
   Asterisk acting as a SIP location server/registrar and use Asterisk 
   as the outbound SIP  proxy. They get calls from Asterisk and place 
   calls to Asterisk. The phone use one of  the SIP domains that are hosted 
   within your Asterisk server. (this is like the  
   current "friend")

   - \b service is when Asterisk is the UA, acting as a phone towards  
   another SIP server - we register with a SIP location server/registrar 
   to get incoming  calls. We place
   calls, masquerading as a phone (using the registrars domain).
   Currently, this is a mixture between a peer (matched on IP for  
   incoming calls) and a register= statement. In some cases, two 
   peers and a register= statement.  Very confusing.

   - \b trunk is when we exchange traffic with another server. We send  
   calls to their SIP domain and receive calls to our SIP domain. We may 
   use realm based authentication for the incoming part of the trunk 
   (not based on caller ID/From: header) and a combination of SIP domain and ACLs.
   This is currently handled by defining sip peers for outbound calls and
   separate SIP peers for inbound calls - where we match on IP. The
   problem with the IP matching is when a trunking partner use several
   SIP servers to connect to us, we need to define one peer per server
   instead of just matching on domain and then authenticate.

*/

/*!
   \page chan_sip3_subs Chan_sip3: Subscriptions

   \title Ideas for a new subscription system
   
   We need to move out the active subsriptions to a list
   of their own, like the registry. Do not keep them in
   the active dialog list, they're active subscriptions.

   Add a list of internal and external subscriptions.
   We need one object that "watches" URIs or extensions
   that is connected to subscribers. For several subscribers,
   we have only one internal or external subscription.

   Should the external subscription system be directly
   connected or go through the hint subsystem? Will this
   cause un-needed overhead?

   exten => johnny,hint,sipsubscribe::sip:johnny@johnnysdomain.com

*/

/*!
   \page chan_sip3_files Chan_sip3: Source code files
   \title Chan_sip3: Source Code Files

   \b \\channels

   - \b chan_sip3.c  The main source code file for the channel
            PBX interface

   \b \\channels\\sip3
   
   - \b sip3.h    The include file for structures and enums
   - \b sip3funcs.h  The include file for functions
   - \b sip3_cliami.c   Manager and CLI functions
   - \b sip3_sdprtp.c   SDP handling and RTP interface
   - \b sip3_callerid.c CallerID, pres and RPID handling
   - \b sip3_monitor.c  The monitor thread (housekeeping)
   - \b sip3_dialog.c   SIP dialog support
   - \b sip3_auth.c  SIP authentication
   - \b sip3_config.c   Configuration
   - \b sip3_domain.c   SIP domain support
   - \b sip3_subscribe.c   SIP subscription support
   - \b sip3_parse.c Parsing stuff
   - \b sip3_compose.c  Composing new requests and responses
   - \b sip3_refer.c SIP transfer and parking support
   - \b sip3_network.c  Networks interface (UDP today)
   - \b sip3_services.c Outbound registrations (services)
   - \b sip3_pokedevice.c  Peer management (health checks)
      
*/
/*!
 * \page chan_sip3_transactions Chan_sip3: Implementing transactions
 *
 * A SIP transaction is a request and one or several responses. 
 * The INVITE transaction is special, it's a three-way handshake
 *    request - response - ack
 * 
 * Responses are divided into temporary responses and final responses.
 * A final response closes the transaction, temporary responses
 * just updates the transaction.
 *
 * Some transactions will open a new SIP dialog, some will not.
 * Some transactions are allowed within an active dialog, some
 * are not.
 *
 * \title   Transactions and chan_sip
 * The current version of chan_sip.c does not have much notion
 *    of a SIP transaction or a dialog. Each transaction will update
 * the dialog, so only the last request is remembered and the
 * direction of the last request.
 * 
 * While this can be seen as effective, it does not allow 
 *    for situations with multiple concurrent transactions.
 * Most importantly, the INVITE transaction can be open for
 * a long time (Ringing) and other things may happen at
 * the same time.
 *
 * We do need to change chan_sip3 so that the transaction
 * state engine works properly.

 *    \title Random thoughts
 * - Integrate sip_pkt and sip_request - done
 * - Mark the UNACKed packets in dialog->packets with a flag
 *    Maybe two lists of packets, to keep it simple for do_monitor?
 *    Or a flag when we have no open transactions?
 * - Use sip-request as "transaction holders" within sip_dialog
 * - Keep them in Cseq order
 * - Add transaction state
 * - When do you delete them - 32 secs after first transmit 
 * - Then we finally can fix the "ignore" ignorant stuff and resend last response
 * - The first transaction that opened in a dialog needs to be saved for the
 *   CANCEL/BYE (in an INVITE transaction)
 * - For SUBSCRIBE dialogs, we need to keep the initial SUBSCRIBE
 *   Then unacknowledged NOTIFY transactions. Keep the transaction
 *   until timer expires
 * - Queue system for incoming packets?
 *
 * SIP_dialog
 *    - sip_trans
 *    Request  (our or remote)
 *    response (our or remote)
 *    - sip_trans
 *
 */
         /* -END- documentation pages */

#define CHAN_SIP3_MAIN

/* GNURK is a temporary marker of functions that are exposed outside of this code file 
   and possibly needs to move out */
#define GNURK

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision: 53128 $")

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

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

#include "sip3/sip3.h"
#include "sip3/sip3funcs.h"

#define SIPLABEL  sip3

/*------- GLOBAL VARIABLES ------------------------------------ */

/*! \brief various expiry times for registrations */
struct expiry_times expiry = {
   .min_expiry = DEFAULT_MIN_EXPIRY,        /*!< Minimum accepted registration time */
   .max_expiry = DEFAULT_MAX_EXPIRY,        /*!< Maximum accepted registration time */
   .default_expiry = DEFAULT_DEFAULT_EXPIRY,
   .expiry = DEFAULT_EXPIRY,     /*!< Is this ever used ??? */
};

/* Default setttings are used as a channel setting and as a default when
   configuring devices */
/*! Global settings only apply to the channel */
struct sip_globals global;

/* Object counters */
struct channel_counters sipcounters = { 0, 0, 0, 0, 0, 0, 0, 0};

/*! \brief Make sure we don't reload twice at the same time */
AST_MUTEX_DEFINE_STATIC(sip_reload_lock);

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 */

struct sched_context *sched;     /*!< The scheduling context */
struct io_context *io;           /*!< The IO context */

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

struct sip_dialog *dialoglist = NULL;        /*!< List of concurrent SIP dialogs */
struct ast_config *notify_types;      /*!< The list of manual NOTIFY types we know how to send */

/*---------------------------- Forward declarations of functions in chan_sip3.c */

/*--- PBX interface functions  - stays in this source code file */
static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause);
static int sip_devicestate(void *data);
static int sip_sendtext(struct ast_channel *ast, const char *text);
static int sip_call(struct ast_channel *ast, char *dest, int timeout);
static int sip_hangup(struct ast_channel *ast);
static int sip_answer(struct ast_channel *ast);
static 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 void receive_message(struct sip_dialog *p, struct sip_request *req);

/*--- Dialog management */
static int auto_congest(void *nothing);
static void list_route(struct sip_route *route);
static void build_route(struct sip_dialog *p, struct sip_request *req, int backwards);
static enum check_auth_result register_verify(struct sip_dialog *p, struct sockaddr_in *sin,
                     struct sip_request *req, char *uri);
static void check_pendings(struct sip_dialog *p);
static int sip_sipredirect(struct sip_dialog *p, const char *dest);

/*--- Codec handling / SDP */
GNURK void try_suggested_sip_codec(struct sip_dialog *p);

/*--- Authentication stuff */
static enum check_auth_result check_user_full(struct sip_dialog *p, struct sip_request *req,
            int sipmethod, char *uri, enum xmittype reliable,
            struct sockaddr_in *sin, struct sip_device **authpeer);
static int check_user(struct sip_dialog *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin);

/*--- Misc functions */
static int sip_sipredirect(struct sip_dialog *p, const char *dest);

/*--- Device monitoring and Device/extension state handling */
static int cb_extensionstate(char *context, char* exten, int state, void *data);
static int sip_devicestate(void *data);

/*--- Applications, functions, CLI and manager command helpers */
GNURK int sip_notify(int fd, int argc, char *argv[]);
static int func_header_read(struct ast_channel *chan, const char *function, char *data, char *buf, size_t len);
static int function_sippeer(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
static int function_sipchaninfo_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
static int sip_dtmfmode(struct ast_channel *chan, void *data);
static int sip_addheader(struct ast_channel *chan, void *data);

/*--- Debugging 
   Functions for enabling debug per IP or fully, or enabling history logging for
   a SIP dialog
*/
GNURK void sip_dump_history(struct sip_dialog *dialog);  /* Dump history to LOG_DEBUG at end of dialog, before destroying data */
GNURK inline int sip_debug_test_pvt(struct sip_dialog *p);

/*--- Device object handling */
static enum parse_register_result parse_register_contact(struct sip_dialog *pvt, struct sip_device *p, struct sip_request *req);

/*--- Parsing SIP requests and responses */
static int get_also_info(struct sip_dialog *p, struct sip_request *oreq);
static int parse_ok_contact(struct sip_dialog *pvt, struct sip_request *req);
static int set_address_from_contact(struct sip_dialog *pvt);
static void check_via(struct sip_dialog *p, struct sip_request *req);
static int get_msg_text(char *buf, int len, struct sip_request *req);

/*--- Constructing requests and responses */
static int create_addr_from_peer(struct sip_dialog *r, struct sip_device *peer);
static int add_vidupdate(struct sip_request *req);

/*------Request handling functions */
static int handle_request_invite(struct sip_dialog *p, struct sip_request *req, int debug, struct sockaddr_in *sin, int *recount, char *e);
static int handle_request_bye(struct sip_dialog *p, struct sip_request *req);
static int handle_request_register(struct sip_dialog *p, struct sip_request *req, struct sockaddr_in *sin, char *e);
static int handle_request_cancel(struct sip_dialog *p, struct sip_request *req);
static int handle_request_message(struct sip_dialog *p, struct sip_request *req);
static int handle_request_subscribe(struct sip_dialog *p, struct sip_request *req, struct sockaddr_in *sin, char *e);
static void handle_request_info(struct sip_dialog *p, struct sip_request *req);
static int handle_request_options(struct sip_dialog *p, struct sip_request *req);
static int handle_invite_replaces(struct sip_dialog *p, struct sip_request *req, int debug, struct sockaddr_in *sin);
static int handle_request_notify(struct sip_dialog *p, struct sip_request *req, struct sockaddr_in *sin, char *e);

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

/*------ T38 Support --------- */
static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_dialog *pvt, int reinvite); /*!< T38 negotiation helper function */

/*! \brief Definition of this channel for PBX channel registration */
static const struct ast_channel_tech sip_tech = {
   .type = "SIP",
   .description = "Session Initiation Protocol (SIP)",
   .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
   .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
   .requester = sip_request_call,      /*!< Where we set up a call, but don't actually activate it */
   .devicestate = sip_devicestate,     /*!< Checking the status of a known SIP device */
   .call = sip_call,       /*!< Try calling Bob, says Alice */
   .hangup = sip_hangup,         /*!< Alice does not want to talk to Bob any more */
   .answer = sip_answer,         /*!< Bob answers the call */
   .read = sip_read,       /*!< Deliver media to the PBX */
   .write = sip_write,        /*!< Get media from the PBX side */
   .write_video = sip_write,     /*!< Get video media from the PBX side */
   .indicate = sip_indicate,     /*!< Get indications from the PBX side */
   .transfer = sip_transfer,     /*!< Transfer a call, severely broken */
   .fixup = sip_fixup,
   .send_digit_begin = sip_senddigit_begin,  /*!< DTMF support */
   .send_digit_end = sip_senddigit_end,      /*!< DTMF support */
   .bridge = ast_rtp_bridge,
   .early_bridge = ast_rtp_early_bridge,
   .send_text = sip_sendtext,    /*!< Get text from the PBX to send out */
};

/*! \brief This version of the sip channel tech has no send_digit_begin
 *  callback.  This is for use with channels using SIP INFO DTMF so that
 *  the core knows that the channel doesn't want DTMF BEGIN frames. */
static const struct ast_channel_tech sip_tech_info = {
   .type = "SIP",
   .description = "Session Initiation Protocol (SIP)",
   .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
   .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
   .requester = sip_request_call,      /*!< Where we set up a call, but don't actually activate it */
   .devicestate = sip_devicestate,     /*!< Checking the status of a known SIP device */
   .call = sip_call,       /*!< Try calling Bob, says Alice */
   .hangup = sip_hangup,         /*!< Alice does not want to talk to Bob any more */
   .answer = sip_answer,         /*!< Bob answers the call */
   .read = sip_read,       /*!< Deliver media to the PBX */
   .write = sip_write,        /*!< Get media from the PBX side */
   .write_video = sip_write,     /*!< Get video media from the PBX side */
   .indicate = sip_indicate,     /*!< Get indications from the PBX side */
   .transfer = sip_transfer,     /*!< Transfer a call, severely broken */
   .fixup = sip_fixup,
   .send_digit_end = sip_senddigit_end,      /*!< DTMF support */
   .bridge = ast_rtp_bridge,
   .early_bridge = ast_rtp_early_bridge,
   .send_text = sip_sendtext,    /*!< Get text from the PBX to send out */
};


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

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


/*! \brief Set nat mode on the various media streams */
GNURK void do_setnat(struct sip_dialog *dialog, int natflags)
{
   const char *mode = natflags ? "On" : "Off";

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

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

   ast_copy_flags(&dialog->flags[0], &device->flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&dialog->flags[1], &device->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
   dialog->capability = device->capability;
   if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && dialog->vrtp) {
      ast_rtp_destroy(dialog->vrtp);
      dialog->vrtp = NULL;
   }
   dialog->prefs = device->prefs;
   if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
      dialog->t38.capability = global.t38_capability;
      if (dialog->udptl) {
         if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_FEC )
            dialog->t38.capability |= T38FAX_UDP_EC_FEC;
         else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY )
            dialog->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
         else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_NONE )
            dialog->t38.capability |= T38FAX_UDP_EC_NONE;
         dialog->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
         if (option_debug > 1)
            ast_log(LOG_DEBUG,"Our T38 capability (%d)\n", dialog->t38.capability);
      }
      dialog->t38.jointcapability = dialog->t38.capability;
   } else if (dialog->udptl) {
      ast_udptl_destroy(dialog->udptl);
      dialog->udptl = NULL;
   }
   do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE );

   if (dialog->rtp) {
      ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
      ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
   }
   if (dialog->vrtp) {
      ast_rtp_setdtmf(dialog->vrtp, 0);
      ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
   }

   /* Set Frame packetization */
   if (dialog->rtp) {
      ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
      dialog->autoframing = device->autoframing;
   }

   /* XXX Why use username for all of this? */
   //ast_string_field_set(dialog, peername, device->defaultuser);

   ast_string_field_set(dialog, peername, device->name);
   /* Authuser is fromuser, and if that not is set, it's the defaultuser, if that's set */
   
   if (ast_strlen_zero(device->extra.fromuser))
      ast_string_field_set(dialog, authname, device->defaultuser);
   else
      ast_string_field_set(dialog, authname, device->extra.fromuser);
   ast_string_field_set(dialog, defaultuser, device->defaultuser);
   ast_string_field_set(dialog, peersecret, device->secret);
   ast_string_field_set(dialog, peermd5secret, device->md5secret);
   ast_string_field_set(dialog, tohost, device->extra.tohost);
   ast_string_field_set(dialog, mohinterpret, device->extra.mohinterpret);
   ast_string_field_set(dialog, mohsuggest, device->extra.mohsuggest);
   ast_string_field_set(dialog, fullcontact, device->fullcontact);
   if (!dialog->initreq && !ast_strlen_zero(device->extra.fromdomain)) {
      char *tmpcall;
      char *c;
      tmpcall = ast_strdupa(dialog->callid);
      c = strchr(tmpcall, '@');
      if (c) {
         *c = '\0';
         ast_string_field_build(dialog, callid, "%s@%s", tmpcall, device->extra.fromdomain);
      }
   }
   if (ast_strlen_zero(dialog->tohost))
      ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
   if (!ast_strlen_zero(device->extra.fromdomain))
      ast_string_field_set(dialog, fromdomain, device->extra.fromdomain);
   if (!ast_strlen_zero(device->extra.fromuser))
      ast_string_field_set(dialog, fromuser, device->extra.fromuser);
   dialog->callgroup = device->callgroup;
   dialog->pickupgroup = device->pickupgroup;
   dialog->allowtransfer = device->allowtransfer;
   /* Set timer T1 to RTT for this peer (if known by qualify=) */
   /* Minimum is settable or default to 100 ms */
   if (device->maxms && device->lastms)
      dialog->timer_t1 = device->lastms < global.t1min ? global.t1min : device->lastms;
   if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
       (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
      dialog->noncodeccapability |= AST_RTP_DTMF;
   else
      dialog->noncodeccapability &= ~AST_RTP_DTMF;
   ast_string_field_set(dialog, context, device->extra.context);
   ast_rtp_set_rtptimeout(dialog->rtp, device->rtptimer.rtptimeout);
   ast_rtp_set_rtpholdtimeout(dialog->rtp, device->rtptimer.rtpholdtimeout);
   ast_rtp_set_rtpkeepalive(dialog->rtp, device->rtptimer.rtpkeepalive);
   if (dialog->vrtp) {
      ast_rtp_set_rtptimeout(dialog->vrtp, device->rtptimer.rtptimeout);
      ast_rtp_set_rtpholdtimeout(dialog->vrtp, device->rtptimer.rtpholdtimeout);
      ast_rtp_set_rtpkeepalive(dialog->vrtp, device->rtptimer.rtpkeepalive);
   }
   if (device->call_limit)
      ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
   dialog->maxcallbitrate = device->maxcallbitrate;
   
   return 0;
}

/*! \brief create address structure from peer name
 *      Or, if peer not found, find it in the global DNS 
 *      returns TRUE (-1) on failure, FALSE on success */
GNURK int create_addr(struct sip_dialog *dialog, const char *username, char *domain, const char *device)
{
   struct hostent *hp;
   struct ast_hostent ahp;
   struct sip_device *p = NULL;
   char *port;
   int portno;
   char host[MAXHOSTNAMELEN], *hostn;
   char peername[256];
   char todomain[256];
   int srvcheck = TRUE;


   dialog->sa.sin_family = AF_INET;
   dialog->timer_t1 = global.t1default; /* Default SIP retransmission timer T1 (RFC 3261) */
   if (!ast_strlen_zero(device)) {
      int res;

      p = find_device(device, NULL, 1);

      if (p)  {
         res = create_addr_from_peer(dialog, p);
         device_unref(p);
         return res;
      } else {
         if (option_verbose > 1)
            ast_verbose("Can't find peer %s\n", device);

         return -1;  /* Can't find peer */
      }
   }

   /* See if we have a port. If we have a given port,
      disable SRV lookups */
   if (!ast_strlen_zero(domain)) {
      ast_copy_string(todomain, domain, sizeof(todomain));
      port = strchr(todomain, ':');
   }
   if (port) {
      *port++ = '\0';
      srvcheck = FALSE;
   }

   hostn = domain;
   portno = port ? atoi(port) : STANDARD_SIP_PORT;

   if (global.srvlookup && srvcheck) {
      char service[MAXHOSTNAMELEN];
      int tportno;
      int ret;

      snprintf(service, sizeof(service), "_sip._udp.%s", domain);
      ret = ast_get_srv(NULL, host, sizeof(host), &tportno, service);
      if (ret > 0) {
         hostn = host;
         portno = tportno;
         if (option_debug > 2)
            ast_log(LOG_DEBUG, "Resolved domain %s to host %s in SRV \n", domain, hostn);
      }
   } 
   hp = ast_gethostbyname(hostn, &ahp);
   if (!hp) {
      ast_log(